Steps Necessary To Pass The 300-815 Exam from Training Expert VCE4Plus
Valid Way To Pass CCNP Collaboration's 300-815 Exam
NEW QUESTION 40
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. 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.
- 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_A User Hold MOH Audio Source settings.
- D. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
- E. 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.
Answer: A,B
NEW QUESTION 41
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. codec g729
- B. incoming called number 555.......
- C. answer-address 555 ........
- D. session-protocol sipv2
Answer: B
NEW QUESTION 42
Refer to the exhibit.
DN 1003 was the last to ring during the most recent call. Which hunting method ensures that DN 1005 is presented with the next call when the hunt pilot is dialed?
- A. parallel
- B. call-blast
- C. sequential
- D. peer
Answer: D
NEW QUESTION 43
Where on Cisco Unified Communications Manager do you configure the standard local route group for a group of devices?
- A. Call Routing > Route/Hunt > Local Route Group Names
- B. Call Routing > Emergency Location > Emergency Location (ELIN) Groups
- C. System > Location Info
- D. System > Device Pool
Answer: A
Explanation:
Reference:
https://www.uccollabing.com/configuring-standard-local-route-group-cucm/
NEW QUESTION 44
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-through calls
- B. interworking between an OOB method and RFC2833 for flow-around calls
- C. interworking between h245-signal and rtp-nte
- D. interworking between h245-alpha numeric and sip-kpml
Answer: B
NEW QUESTION 45
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.
Answer: D
NEW QUESTION 46
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. Add the long-distance & international pattern's partitions to the calling search space of the physical phone.
- B. Remove the long-distance & international pattern's partitions from the calling search space of the device profile.
- C. Add the long-distance & international pattern's partitions to the calling search space of the device profile
- D. Remove the long-distance & international pattern's partitions from the calling search space of the physical phone.
- E. Add the long-distance & international pattern's partitions to the calling search space of the physical phone's directory number.
Answer: C,D
NEW QUESTION 47
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. Cisco IP Manager Assistant
- B. CallManager traces
- C. CTI Manager traces
- D. Call logs
Answer: B
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION 48 
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.
Answer: D
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION 49
An engineer is configuring a call park feature in Cisco Unified Communications Manager Express. Which command does the engineer use to ensure that the call is reverted to the user after 60 seconds?
- A. R2(config-ephone-dn)#park-slot timeout 30 limit 2 recall alternate 3002
- B. R2(config-ephone-dn)#park reservation-group 1
- C. R2(config-ephone-dn)#park-slot timeout 60 limit 2 recall alternate 3002
- D. R2(config-ephone-dn)#park reservation-group 60
Answer: A
NEW QUESTION 50
How does an engineer globalize routing for ingress calls coming from the PSTN to internal DNs?
- A. At the PSTN gateway, put the calling number in PSTN format and the called number in DN format.
- B. At Cisco Unified CM, put the calling number in E.164 format and the called number in PSTN format.
- C. At Cisco Unified Communications Manager, put the calling number in E.164 format and the called number in E.164 format.
- D. At the PSTN gateway, put the calling number in E.164 format and the called number in localized (DN) format.
Answer: B
NEW QUESTION 51
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 has the option to configure Authorization Level such as in the Forced Authorization Code.
- 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.
- C. If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.
- D. If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.
- E. 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.
Answer: B,C
NEW QUESTION 52
Which description of RTP timestamps or sequence numbers is true?
- A. Sequence numbers increase by four for each RTP packet transmitted.
- B. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout
- C. Timestamps increase by the time "carrying" by a packet.
- D. The sequence number is used to detect losses.
Answer: B
Explanation:
delay compensation).
NEW QUESTION 53
What are the elements for Device Mobility configuration?
- A. device pool, Device Mobility group, and region
- B. device pool, Device Mobility group, and Cisco IP phone
- C. physical location, device pool, and Device Mobility group
- D. physical location. Device Mobility group, and region
Answer: C
Explanation:
Reference:
https://www.ciscopress.com/articles/article.asp?p=1249228&seqNum=4
NEW QUESTION 54
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. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.
- C. Early Offer for G Clear Calls must be enabled.
- D. Accept Audio Codec Preferences in Received Offer must be set to On.
Answer: B
NEW QUESTION 55
The Cisco Unified Communications Manager Dialed Number Analyzer allows analysis of calls from which two devices? (Choose two.)
- A. device pools
- B. CTI route points
- C. IP phones
- D. translation patterns
- E. CTI ports
Answer: C,E
Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/dna/11_5_1/ CUCM_BK_CBA47A6E_00_cucm-dna-guide-115/CUCM_BK_CBA47A6E_00_cucm-dna-guide- 115_chapter_01.html#CUCM_TP_A5DA99E0_00
NEW QUESTION 56
Refer to the exhibit.
How many maximum hops can an ILS update traverse?
- A. 0
- B. 1
- C. 2
- D. 3
Answer: B
NEW QUESTION 57
An administrator sees the voice register pool 1 command in a Cisco UCME configuration. Which configuration is occurring in this section?
- A. configuration items common for all SIP phones
- B. configuration for SIP registrar service
- C. configuration for a single SIP phone
- D. configuration for a pool of SIP phones (similar to device pool on Cisco UCM)
Answer: C
NEW QUESTION 58
If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern "123"?
- A. 12[2-5]
- B. 12!
- C. 1XX (urgent Priority Set)
- D. 12X (urgent priority set)
Answer: A
NEW QUESTION 59
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 asn 1
- B. debug H.323 messages
- C. debug H.225 asn1
- D. debug H.246 asn 1
- E. debug H.225 media
Answer: C
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION 60
An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which two debugs must the administrator turn on? (Choose two).
- A. debug H.323 asn1
- B. debug H.245 asn1
- C. debug H.323 message
- D. debug H.225 asn1
- E. debug H.225 media
Answer: B,D
NEW QUESTION 61
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 Security Information of the SIP Profile configuration.
- B. Incoming Port in SIP Information section of the SIP Trunk configuration.
- C. Destination Port in SIP Trunk Security Profile configuration
- D. Destination Port in SIP Information section of the SIP Trunk configuration
- E. Incoming Port in SIP Trunk Security Profile configuration
Answer: D,E
NEW QUESTION 62 
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. 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.
- 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_A User Hold MOH Audio Source settings.
- D. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.
- E. 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.
Answer: A,B
Explanation:
Section: Signaling and Media Protocols
NEW QUESTION 63
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?
- A. voice-class sip resources priority mode passthrough
- B. voice-class sip bind control source-interface Loopback0
- C. voice-class sip bind media source-interface Loopback0
- D. voice-class sip early-offer forced.
Answer: C
NEW QUESTION 64
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