Free Cisco 300-815 CLACCM Practice Questions 2026 Part1

Question #1

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.)

A. "voice register pool " configuration mode

B. "ephone-dn " configuration mode

C. "dial-peer cor custom " configuration mode

D. "voice register global " configuration mode

E. "telephony-service " configuration mode

View answer

Correct Answer: AB

Question #2

A support engineer is troubleshooting a voice network.When conducting a search for call setup details related to calling search space issues, which trace files should be investigated?

A. CallManager traces

B. CTI Manager traces

C. Cisco IP Manager Assistant

D. Call logs

View answer

Correct Answer: A

Question #3

Refer to the exhibit.Users report that outbound PSTN calls from phones registered to Cisco Unified Communications Manager are not completing. The local service provider in North America has a requirement to receive calls in 10-digit format. The Cisco Unified CM sends the calls to the Cisco Unified Border Element router in a globalized E.164 format. There is an outbound dial peer on Cisco Unified Border Element configured to send the calls to the provider. The dial peer has a voice translation profile applied in the correct direction but an incorrect voice translation rule applied, which is shown in the exhibit. Which rule modified DNIS in the format that the provider is expecting?

A. ule 1 /^\([2-9]

B. ule 1/^\+1\([2-9]

C. ule 1 /^\+1\([2-9]

D. ule 1 /^/+\([^1]

View answer

Correct Answer: B

Question #4

A user presses the Messages button but is not redirected to voicemail. Direct dialing of the voicemail pilot works. What is the most likely cause?

A. Voicemail pilot is not in the CSS

B. Voicemail profile is not assigned to the phone

C. Unity Connection service is stopped

D. SIP trunk is misconfigured

View answer

Correct Answer: B

Question #5

Users of a Cisco UCME report that calls lose audio after they are forwarded to some users. Calling point-to- point seems to function fine, but depending on the type of phone, some calls lose audio when forwarded.Looking at the trace header that shows Allow:INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTERWhat is missing and is a reason that the SRTP stream stops working?

A. EARLY-OFFER

B. FORWARD

C. UPDATE

D. REDIRECT

View answer

Correct Answer: C

Question #6

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.)

A. three-way conference

B. secure SIP lines

C. T

D. transcoding

E. SIP trunk

View answer

Correct Answer: AC

Question #7

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

View answer

Correct Answer: C

Question #8

Refer to the exhibit. Regions have been configured for all major branches based on the available circuit bandwidth. Some calls from Region A endpoints to Region B endpoints are failing to connect. How is the issue resolved?

A. Update the calling search space for affected endpoints to none

B. Update all regions to 8 kbps maximum audio bitrate

C. Increase the number of available media termination points

D. Add a media resource to transcode between available capabilities

View answer

Correct Answer: D

Question #9

Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. Allcalls 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

View answer

Correct Answer: AC

Question #10

Which description of RTP timestamps or sequence numbers is true?

A. he sequence number is used to detect losses

B. imestamps increase by the time “carrying” by a packet

C. equence numbers increase by four for each RTP packet transmitted

D. he timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation)

View answer

Correct Answer: D

Question #11

Refer to the exhibit. An administrator received reports that users do not hear a ringback when they dial specific external numbers. For example, the Cisco Unified Border Element receives a delayed offer invite from Cisco UCM and responds with the 183 session progress in the exhibit. Assume that the PSTN is responsible for generating ringback in these problematic scenarios.Which action resolves the issue?

A. On Cisco Unified Border Element:voice service voip sip rel1xx disable 100rel

B. Edit the configuration for SIP Rel1XX Options in the appropriate SIP profile on Cisco UCM to Send PRACK if 1xx Contains SDP

C. On Cisco Unified Border Element:voice service voip sip rel1xx require 100rel

D. Enable early media on the appropriate SIP trunk on Cisco UCM

View answer

Correct Answer: B

Question #12

When an administrator troubleshoots H.323 call setup, which message gives an alert that the called party is being notified about the call?

A. ALERTING

B. PROCEEDING

C. CONNECT

D. RINGING

View answer

Correct Answer: C

Question #13

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.)

A. three-way conference

B. secure SIP lines

C. T

D. transcoding

E. SIP trunk

View answer

Correct Answer: AC

Question #14

Where on Cisco Unified Communications Manager do you configure the standard local route group for a group of devices?

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

View answer

Correct Answer: B

Question #15

When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?

A. ALERTING

B. PROCEEDING

C. CONNECT

D. RINGING

E. Reveal Answer

View answer

Correct Answer: A

Question #16

An IP Telephony administrator is deploying IP phones. The administrator has an existing Cisco UCME router with several SCCP & SIP phones registered. The administrator receives a request for a new SIP phone with MAC address 1111.2222.3333 and directory number 2050 to be added in the Cisco UCME. Which two configurations should be added in CME to support this request? (Choose two.)

A.

B.

C.

D.

E.

View answer

Correct Answer: DE

Question #17

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?

A. voice service voip enable ip address trust list

B. voice service voip ip address trusted list

C. voice service voip ip address trusted authenticate

D. voice service voip enable ip address trust authentication

View answer

Correct Answer: B

Question #18

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. It is determined 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 solutions for this issue? (Choose two.)

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

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

View answer

Correct Answer: AC

Question #19

55697959.007 |12:20:50.913 |AppInfo |RouteListCdrc::createPartyTransformedCcSetupReqMsg - before DAapplyCdpnXform() preXformCdpn=11112222 preTag=SUBSCRIBER prePos=11112222 crCdpnMask=33334444 crPrefixDigit= crDDI=255697959.008 |12:20:50.913 |AppInfo |RouteListCdrc::createPartyTransformedCcSetupReqMsg - after DAapplyCdpnXform() xformCdpn=33334444 xformTag=SUBSCRIBER xformPos=1111222255697959.009 |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

B. 55698034

C. 55698034

D. 55698034

View answer

Correct Answer: C

Question #20

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.)

A. Create one global route list for PSTN calls that points to one global PSTN route group

B. Create a route group which has all the gateways and associate it to the device pool of every site

C. Assign one route group as a local route group in the device pool of the corresponding site

D. Create one route group for each site and one global route list for PSTN calls that point to the local route group

E. Create a hunt group and assign it to each side route pattern

View answer

Correct Answer: AC

Question #21

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.)

A. three-way conference

B. secure SIP lines

C. T

D. transcoding

E. SIP trunk

View answer

Correct Answer: AC

Question #22

An engineer is configuring call queuing to place callers in a queue until hunt members are available to answer them. The engineer also must set the option to automatically log off an agent that does not answer a queuing-enabled hunt pilot call.Which Cisco UCM menu is used to accomplish this goal?

A. Call Routing > Route/Hunt > Line Group

B. Call Routing > Route/Hunt > Hunt Group

C. Call Routing > Route/Hunt > Hunt Pilot

D. Call Routing > Route/Hunt > Call Queuing

View answer

Correct Answer: C

Question #23

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.)

A. Remove the long-distance and international pattern's partitions from the calling search space of the physical phone

B. Add the long-distance and international pattern's partitions to the calling search space of the physical phone's directory number

C. Remove the long-distance and international pattern's partitions from the calling search space of the device profile

D. Add the long-distance and international pattern's partitions to the calling search space of the physical phone

E. Add the long-distance and international pattern's partitions to the calling search space of the device profile

View answer

Correct Answer: AE

Question #24

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.)

A. Create one global route list for PSTN calls that points to one global PSTN route group

B. Create a route group which has all the gateways and associate it to the device pool of every site

C. Assign one route group as a local route group in the device pool of the corresponding site

D. Create one route group for each site and one global route list for PSTN calls that point to the local route group

E. Create a hunt group and assign it to each side route pattern

View answer

Correct Answer: AC

Question #25

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-vers a. As an administrator, which additional configuration should be made to fulfill the user's request?

A. dd the mobility key to the softkey template that the desk phone is using

B. heck to make sure that the Resume softkey option appears on the desk phone

C. se Dialed Number Analyzer to determine if the user extension can dial the mobile phone

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

View answer

Correct Answer: A

Question #26

Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)

A. TMF

B. FCP

C. IDEO

D. AX

E. UDIO

View answer

Correct Answer: AD

Question #27

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.)

A. three-way conference

B. secure SIP lines

C. T

D. transcoding

E. SIP trunk

View answer

Correct Answer: AC

Question #28

Refer to the exhibit.In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone user C. Which two scenarios are correct? (Choose two.)

A. Phone_A sends a SIP - REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer - To section

B. Phone_B sends a SIP - REFER message to the Cisco Unified CM with Phone_C information in the Refer - To section

C. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_B User Hold MOH Audio Source settings

D. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings

E. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_A User Hold MOH Audio Source settings

View answer

Correct Answer: AD

Question #29

An administrator is troubleshooting a one-way audio issue for a call that uses H.323 protocol in slow-start mode. The administrator requests that the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audio call is provided. The H.225 and H.245 messages for one of the one-way audio calls are gathered and the call flow has not invoked any media resources.Where is the RTP IP and port information for both sides found?

A. H

B. H

C. H

D. H

View answer

Correct Answer: B

Question #30

A company is using Cisco Jabber on-premises to make B2B calls on video. The calls are using Cisco Expressway-C and Expressway-E and have been configured in Cisco UCM to be able to call any URI on the internet. The Jabber client also has voice enabled and must be able to call local, regional, and international numbers. Where must Cisco UCM be configured to meet this requirement for URI dialing?

A. nter "

B. nter "!#" in the SIP route pattern

C. nter "!#" in the route pattern section tied to a route group and list

D. nter "*" in the SIP route pattern

View answer

Correct Answer: A

Question #31

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?

A. voice service voip enable ip address trust list

B. voice service voip ip address trusted list

C. voice service voip ip address trusted authenticate

D. voice service voip enable ip address trust authentication

View answer

Correct Answer: B

Question #32

An engineer receives reports from users that calls to external participants on mobile phones are failing. The gateway is based on a server client setup.When logs are analyzed, what is the last message expected before the media is established that Cisco UCM sends?

A. established connection with an early SDP to the gateway, which must respond with 200 OK

B. modified connection with SDP to the gateway, which must respond with 200 OK

C. established connection with SDP to the gateway, which must respond with 200 OK

D. forward connection with an early offer to the gateway, which must respond with 200 OK

View answer

Correct Answer: C

Question #33

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

C. The rtpmap0 value for the negotiated codec is marking DTMF as inactive

D. No DTMF is negotiated

View answer

Correct Answer: D

Question #34

Where is the dtmf-relay command configured on Cisco Unified Border Element?

A. Ain the voice-class VoIP configuration

B. Bin the VoIP dial peer

C. Cin global SIP configuration

D. Din the VoIP or POTS dial peers

View answer

Correct Answer: B

Question #35

Refer to the exhibit.For long-distance calls, users must prefix their dialed number with "91." The translation pattern was created to strip the 91 as the PSTN expects a 10- digit number. The PSTN also requires the calling number to be set to 9195551234. However, the service provider has said calls with a different calling number are being received. How is this issue resolved?

A. hange the partition of the translation pattern from none to pstn_pt

B. nable Force Authorization Code on the route pattern

C. nable Use Calling Party's External Phone Number Mask on the translation pattern

D. isable Use Calling Party's External Phone Number Mask on the route pattern

View answer

Correct Answer: D

Question #36

An engineer is troubleshooting local ringback on a Cisco SIP gateway. The gateway is not ignoring the SIP 180 response with SDP from the service provider, and the far end device is sending the 180 with SDP to play ringback from the IP address specified in the SDP.Which configuration change must be made on the gateway to resolve the issue?

A. Router(config-sip-ua)# no disable-early-media 180

B. Router(conf-voi-serv)# no disable-early-media 180

C. Router(conf-voi-serv)# disable-early-media 180

D. Router(config-sip-ua)# disable-early-media 180

View answer

Correct Answer: D

Question #37

Refer to the exhibit. An engineer is troubleshooting an issue with the caller not hearing a PSTN announcement before the SIP call has completed setup. How must the engineer resolve this issue using the reliable provisional response of the SIP?

A. voice service voip sip no rel1xx

B. sip-ua disable-early-media 180

C. voice service voip sip rel1xx require 100rel

D. voice service voip sip send 180 sdp

View answer

Correct Answer: C

Question #38

What are two functions of the metadata in SIP recording? (Choose two.)

A. includes the details of the application media stream

B. contains the cryptographic context of the SRTP passthrough calls

C. identifies the session and media association time

D. carries the Agent Statistics Report

E. carries the communication session data (audio and video calls) that describes the call to the recording server

View answer

Correct Answer: CE

Question #39

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?

A. voice service voip enable ip address trust list

B. voice service voip ip address trusted list

C. voice service voip ip address trusted authenticate

D. voice service voip enable ip address trust authentication

View answer

Correct Answer: B

Question #40

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?

A. voice service voip enable ip address trust list

B. voice service voip ip address trusted list

C. voice service voip ip address trusted authenticate

D. voice service voip enable ip address trust authentication

View answer

Correct Answer: B

Question #41

An engineer is troubleshooting local ringback on a Cisco SIP gateway. The gateway is not ignoring the SIP 180 response with SDP from the service provider, and the far end device is sending the 180 with SDP to play ringback from the IP address specified in the SDP.Which configuration change must be made on the gateway to resolve the issue?

A. Router(config-sip-ua)# no disable-early-media 180

B. Router(conf-voi-serv)# no disable-early-media 180

C. Router(conf-voi-serv)# disable-early-media 180

D. Router(config-sip-ua)# disable-early-media 180

View answer

Correct Answer: D

Question #42

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?

A. voice service voip enable ip address trust list

B. voice service voip ip address trusted list

C. voice service voip ip address trusted authenticate

D. voice service voip enable ip address trust authentication

View answer

Correct Answer: B

Question #43

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).

A. H

B. H

C. H

D. H

View answer

Correct Answer: B

Question #44

In Cisco UCM, which tool is used to check SIP traces?

A. TP

B. CSIP

C. S Administration Page

D. TMT

View answer

Correct Answer: D

Question #45

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

View answer

Correct Answer: C

Question #46

A company is using a H.323 gateway for PSTN calls from Cisco phones connected to a Cisco UCM cluster. There are reports of problems getting audio on these incoming calls from the PSTN. The phone rings, but nothing is heard and the call disconnects.Which command troubleshoots this issue?

A. debug h245 asn1

B. debug h245 pstn

C. debug voice h323 inout

D. debug voice sip inout

View answer

Correct Answer: C

Question #47

Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?

A. Analysis Manager > Inventory > Trace File Repositories

B. System > Tools > Trace and Log Central

C. Voice/Video > Session Trace Log View > Real Time Data

D. Voice/Video > Session Trace Log View > Open From Local Disk

View answer

Correct Answer: C

Question #48

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

View answer

Correct Answer: C

Question #49

Refer to the exhibit. In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone user C. Which two scenarios arecorrect? (Choose two.)

A. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section

B. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section

C. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_B User Hold MOH Audio Source settings

D. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings

E. As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_A User Hold MOH Audio Source settings

View answer

Correct Answer: AD

Question #50

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.)

A. three-way conference

B. secure SIP lines

C. T

D. transcoding

E. SIP trunk

View answer

Correct Answer: AC

Question #51

An engineer is troubleshooting local ringback on a Cisco SIP gateway. The gateway is not ignoring the SIP 180 response with SDP from the service provider, and the far end device is sending the 180 with SDP to play ringback from the IP address specified in the SDP.Which configuration change must be made on the gateway to resolve the issue?

A. Router(config-sip-ua)# no disable-early-media 180

B. Router(conf-voi-serv)# no disable-early-media 180

C. Router(conf-voi-serv)# disable-early-media 180

D. Router(config-sip-ua)# disable-early-media 180

View answer

Correct Answer: D

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Free Cisco 300-815 CLACCM Practice Questions 2026 Part1

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