Cisco

300-815 Exam
CCNP

Questions & Answers

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Version: 11.1

Question: 1

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

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MOH audio is chosen from Phone_A User Hold MOH Audio Source settings.

Answer: AD
Explanation:

Question: 2

Refer to the exhibit.

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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.

Answer: D
Explanation:

Question: 3

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.245 Terminal Capability Set

B. H.245 Open Logical Channel

C. H.225 Connect

D. H.245 Open Logical Channel Ack

Answer: B
Explanation:

Reference: http://ccievoicehopeful.blogspot.com/2012/09/h323-notes.html

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Question: 4

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. DTMF

B. BFCP

C. VIDEO

D. FAX

E. AUDIO

Answer: AB
Explanation:

Question: 5

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

Answer: A
Explanation:

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Question: 6

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

Answer: D
Explanation:

Question: 7

Why would RTP traffic that is sent from the originating endpoint fail to be received on the far
endpoint?

A. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call
signaling path.

B. Cisco Unified Communications Manager invoked media termination point resources.

C. The RTP traffic is arriving beyond the jitter buffer on the receiving end.

D. A firewall in the media path is blocking TCP ports 16384-32768.

Answer: C
Explanation:

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Question: 8

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

Answer: B
Explanation:

Question: 9

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

Answer: A
Explanation:

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Reference: https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-
voip/211306-In- Depth-Explanation-of-Cisco-IOS-and-IO.html#anc8

Question: 10

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.

Answer: AC
Explanation:

Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/port/9_1_1/
CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-91/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-
91_chapter_01.html

Question: 11

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

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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

Answer: C
Explanation:

Reference: https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-
communications- manager-callmanager/213583-procedure-to-analyse-call-flow-of-sip-ca.html

Question: 12

Which description of RTP timestamps or sequence numbers is true?

A. The sequence number is used to detect losses.

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

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

D. The timestamp is used to place the incoming audio and video packets in the correct timing order
(playout

delay compensation).

Answer: D
Explanation:

Reference: https://www.cs.columbia.edu/~hgs/rtp/faq.html

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Question: 13

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

Answer: A
Explanation:

Question: 14

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

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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.

Answer: C
Explanation:

Question: 15

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.38 fax relay

D. transcoding

E. SIP trunk

Answer: AC
Explanation:

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Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/
guide/SCCP_and_SIP_SRST_Admin_Guide/srst_sip_isr4000.html

Question: 16

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”.

D. Enable Secondary Dial tone on Analog and Digital FXO Ports.

Answer: B
Explanation:

Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/manual/
cmeadm/cmetoll.html#concept_ECC4F4E7ED0F45C594B703EEF34762F2

Question: 17

You see the voice register pool 1 command in your Cisco Unified Communications Manager Express
configuration. Which configuration is occurring in this section?

A. configuration for a single SIP phone

B. configuration items common for all SIP phones

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C. configuration for a pool of SIP phones (similar to device pool on Cisco Unified Communications
Manager)

D. configuration for SIP registrar service

Answer: A
Explanation:

Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/
guide/SCCP_and_SIP_SRST_Admin_Guide/srst_setting_up_using_sip.html

Question: 18

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?

A. allow-connections sip to sip

B. voice service voip

C. voice register global

D. voice register dn

Answer: B
Explanation:

Reference: https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-
communications-manager-express/99946-cme-sip-guide.html

Question: 19

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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

Answer: A
Explanation:

Reference: https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border-
element/200412-DTMF-Relay-and-Interworking-on-CUBE.html#anc35

Question: 20

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

Answer: B
Explanation:

Reference: https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-
book/dtmf- relay.html

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