Question 1

Where is the dtmf-relay command configured on Cisco Unified Border Element?
A) in the voice-class VoIP configuration
B) in the VoIP dial peer
C) in global SIP configuration
D) in the VoIP or POTS dial peers

Accepted Answer

The dtmf-relay command is configured in the VoIP dial peer to specify how DTMF tones are relayed across the network.

Question 2

Refer to the exhibit. An administrator is troubleshooting why users are not hearing audio when dialing long distance numbers across their Cisco Unified Border Element. The customer's carrier has a requirement that dialing long distance requires an access code to be entered. Looking at the exhibit, what two actions can be taken to correct signaling? (Choose two.)
A) Enanle PRACK.
B) Enable Early Offer on the Cisco Unified Border Element.
C) Enable the supplementary-service media-renegotiate command. Enable the supplementary-service media-renegotiate command.
D) Enable Media Flow Around
E) Enable Mid-Call Signaling Consumption.

Accepted Answer

A,B

Question 3

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

Accepted Answer

The first preference condition matched in a SIP-enabled incoming dial peer is the "incoming uri".

Question 4

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

Accepted Answer

CallManager traces should be investigated for call setup details related to calling search space issues, as they contain detailed logs of call processing and signaling events.

Question 5

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.

Accepted Answer

The answer of  Refer to the exhibit. Users report...

Question 6

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

Accepted Answer

The answer of A user in location X dials an...

Question 7

Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?
A) Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.
B) Configure IP Address Trusted Authentication for Incoming VoIP Calls.
C) Configure the command no ip address trusted authenticate under "voice service voip". Configure the command no ip address trusted authenticate under "voice service voip".
D) Enable Secondary Dial tone on Analog and Digital FXO Ports.

Accepted Answer

Configuring IP Address Trusted Authentication for Incoming VoIP Calls helps prevent unauthorized access and toll fraud by ensuring that only calls from trusted IP addresses are accepted.

Question 8

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.)
A) example.com@gateway.corporate.local
B) *@example.com
C) gateway.corporate.local
D) example.com
E) *.*

Accepted Answer

The SIP route pattern *@example.com matches any SIP address with the domain example.com, and the SIP route pattern *.* matches any SIP address.

Question 9

Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)
A) can be associated under the route group
B) can be associated only under the route list
C) chooses the route group that is configured under the device pool of the calling-party device
D) chooses the route group that is configured under the device pool of the called-party device
E) can be assigned directly to the route pattern

Accepted Answer

The answer of Which two descriptions of the Standard Local...

Question 10

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

Accepted Answer

The answer of When you troubleshoot H.323 call setup, which...

Question 11

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

Accepted Answer

The command "dial-peer voice 20 voip" is used to create a VoIP dial peer, which can be SIP-enabled. SIP is a protocol used for VoIP communications.

Question 12

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 Contact: header of the 200 OK response
B) Allow: header if the 200 OK response Allow: header if the 200 OK response
C) o= line of SDP content o= line of SDP content
D) c= line of SDP content c= line of SDP content

Accepted Answer

The answer of End users at a new site report...

Question 13

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 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 Router(config)# dial-peer voice 2 voip Router(config-dial-peer)# no vad
C) Router(conf-voi-serv)# allow-connections voice-mail mod allow-connections voice-mail mod
D) Router(conf-voi-serv)# no supplementary-service sip moved-temporarily no supplementary-service sip moved-temporarily

Accepted Answer

The answer of After configuring a Cisco CallManager Express with...

Question 14

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

Accepted Answer

The pattern 813510[012345] specifically matches numbers 8135100 through 8135105 by listing each digit in the range.

Question 15

An administrator is troubleshooting call failures on an H323 gateway via the CLI. To see signaling for media and call setup, which debug must the Administrator turn on?
A) debug H323 messages
B) debug H225 asn1
C) debug H246 asn 1
D) debug H225 media
E) debug H323 asn 1

Accepted Answer

The answer of An administrator is troubleshooting call failures on...

Question 16

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.

Accepted Answer

The Client Matter Code feature does not support overlap sending because it cannot determine when to prompt the user for the code, and enabling overlap sending disables the requirement for a Client Matter Code.

Question 17

Refer to the exhibit. A user reports that when they call a specific phone number, no one answers the call, but when they call from a mobile phone, the call is answered. The engineer troubleshooting the issue is expecting the far-end gateway to cut through audio on the 183 Session Progress SIP message. Which SIP Profile configuration element is necessary for the Cisco Unified Communications Manager to send acknowledgement of provisional responses?
A) Allow Passthrough of Configured Line Device Caller Information must be enabled.
B) Accept Audio Codec Preferences in Received Offer must be set to On.
C) On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.
D) Early Offer for G Clear Calls must be enabled.

Accepted Answer

The answer of  Refer to the exhibit. A user...

Question 18

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.)
A) Incoming Port in SIP Information section of the SIP Trunk configuration.
B) Incoming Port in Security Information of the SIP Profile configuration.
C) Destination Port in SIP Information section of the SIP Trunk configuration
D) Incoming Port in SIP Trunk Security Profile configuration
E) Destination Port in SIP Trunk Security Profile configuration

Accepted Answer

The answer of An engineer must configure a secure SIP...

Question 19

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

Accepted Answer

The "Voice/Video > Session Trace Log View > Real Time Data" section is specifically designed for reviewing the call flow and signaling for a SIP call in real time.

Question 20

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) rule 1 /^/+$[^1].*$/ /011\1/
B) rule 1/^\+1$[2-9]..[2-9]……$$/ /\1/
C) rule 1 /^$[2-9]..[2-9]……$$/ /\1/
D) rule 1 /^\+1$[2-9]..[2-9]……$$/ /\0/

Accepted Answer

The answer of  Refer to the exhibit. Users report...

Exam 30: Implementing Cisco Collaboration Cloud and Edge Solutions

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

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

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?

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 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?

Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?

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.)

Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)

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

Which IOS command creates a SIP-enabled dial peer?

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?

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 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?

An administrator is troubleshooting call failures on an H323 gateway via the CLI. To see signaling for media and call setup, which debug must the Administrator turn on?

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

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.)

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