[Q38-Q63] 100% Free 300-815 Exam Dumps Use Real CCNP Collaboration Dumps With 120 Questions!

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100% Free 300-815 Exam Dumps Use Real CCNP Collaboration Dumps With 120 Questions!

Pass Your 300-815 Exam Easily With 100% Exam Passing Guarantee [2022]

NEW QUESTION 38
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. +E.164 alternate number mask
  • D. external phone number mask
  • E. enterrise alternate number mask

Answer: A,D

Explanation:
Reference:
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/dialplan.html

 

NEW QUESTION 39

Refer to the exhibit. Users report that when they dial the emergency number 9911 from any internal phone, it takes a long time to connect with the emergency operator. Which action resolves this issue?

  • A. Adjust the service parameter T204 timer to the desired value.
  • B. Check the Urgent Priority check box under 9.911 pattern.
  • C. Point the emergency pattern directly to the PSTN gateway.
  • D. Adjust the service parameter T302 timet to the desired value.

Answer: B

Explanation:
Section: Call Control and Dial Planning

 

NEW QUESTION 40
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. peer
  • D. sequential

Answer: C

 

NEW QUESTION 41
Refer to the exhibit.

While troubleshooting call failures on the Cisco Unified Border Element, an administrator notices that messages are being sent to the service provider, but there is no response The administrator later learns that this SIP provider does not support PRACK. Which header should be removed from the SIP message to resolve this issue?

  • A. Require 100rel
  • B. Contact <sip:[email protected]:5060>
  • C. Content-Disposition: session:handling=required
  • D. Content-Type: application/sdp

Answer: A

 

NEW QUESTION 42
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 > Device Pool
  • D. System > Location Info

Answer: A

Explanation:
Section: Call Control and Dial Planning
Explanation/Reference: https://www.uccollabing.com/configuring-standard-local-route-group-cucm/

 

NEW QUESTION 43
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 60 limit 2 recall alternate 3002
  • B. R2(config-ephone-dn)#park-slot timeout 30 limit 2 recall alternate 3002
  • C. R2(config-ephone-dn)#park reservation-group 60
  • D. R2(config-ephone-dn)#park reservation-group 1

Answer: B

 

NEW QUESTION 44
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. Call logs
  • C. CallManager traces
  • D. CTI Manager traces

Answer: C

Explanation:
Section: Signaling and Media Protocols

 

NEW QUESTION 45
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 > Device Pool
  • D. System > Location Info

Answer: A

Explanation:
Reference:
https://www.uccollabing.com/configuring-standard-local-route-group-cucm/

 

NEW QUESTION 46
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. No DTMF is negotiated.
  • B. There is SIP Delayed Offer. DTMF is supported only in Early Offer.
  • C. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
  • D. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.

Answer: A

 

NEW QUESTION 47
An administrator configured Cisco Unified Mobility to block access to remote destinations for certain caller IDs. A user reports that a blocked caller was able to reach a remote destination. Which action resolves the issue?

  • A. Configure a mobility identity.
  • B. Configure Single Number Reach.
  • C. Configure Mobile Voice Access.
  • D. Configure an access list.

Answer: D

 

NEW QUESTION 48
Refer to the exhibit.

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

  • A. rulel /.*\(3548S\)/^1/
  • B. rulel /^9\(\d*\)/^1/
  • C. rule 1 /^9\(.*\)/A1/
  • D. rule 1/^9123548/^1/

Answer: C

 

NEW QUESTION 49
Refer to the exhibit.

Within the North American Numbering Plan, gateways located in Ottawa, Canada and marked as "YOW" are assigned to the Calling Party Transformation CSS NANP_CgPTP, which contains partition NANP_calling_xforms. What is the calling-party number and the numbering type if the calling user
+1613-555-1234 dials the number?

  • A. calling number 613-555-1234 and numbering type "subscriber"
  • B. calling number 011-1-613-555-1234 and numbering type "subscriber"
  • C. calling number 011613-555-1234 and numbering type "international"
  • D. calling number 613-555-1234 and numbering type "national"

Answer: D

 

NEW QUESTION 50
When a third-party SIP Phone System is dialed inbound across a Cisco Unified Border Element, DTMF is failing. The third-party vendor accepts only out-of-band DTMF. Which configuration should be added to the outgoing dial peer to resolve this issue?

  • A. dtmf-relay sip-kpml
  • B. dtmf-relay cisco-rtp
  • C. dtmf-relay rtp-nte
  • D. dtmf-relay h245-signal

Answer: A

 

NEW QUESTION 51
If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern "123"?

  • A. 1XX (urgent Priority Set)
  • B. 12X (urgent priority set)
  • C. 12[2-5]
  • D. 12!

Answer: B

Explanation:
Section: Call Control and Dial Planning

 

NEW QUESTION 52
Refer to the exhibit.

A Cisco Unified Border Element continues to send 180/183 with the required: 100rel header to Cisco UCM. and the call eventually disconnects How is the issue resolved?

  • A. Enable *Early Offer support for voice and video calls" on the SIP Profile Configuration Page in Cisco UCM.
  • B. Disable "SIP Rel1XX Options* and 'Early Offer Support* on the SIP Profile Configuration Page m Cisco UCM.
  • C. Disable "Send send-receive SDP in mid-call INVITE* on the SIP Profile Configuration Page in Cisco UCM.
  • D. Enable 'SIP ReI1XX Options* and -Early Offer Support" on the SIP Profile Configuration Page in Cisco UCM.

Answer: A

 

NEW QUESTION 53
Which description of RTP timestamps or sequence numbers is true?

  • A. The sequence number is used to detect losses.
  • B. Sequence numbers increase by four for each RTP packet transmitted.
  • C. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
  • D. Timestamps increase by the time "carrying" by a packet.

Answer: C

Explanation:
Section: Signaling and Media Protocols
Explanation/Reference: https://www.cs.columbia.edu/~hgs/rtp/faq.html

 

NEW QUESTION 54
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. 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.
  • B. Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager 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 music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings.
  • D. 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.
  • E. Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.

Answer: B,C

 

NEW QUESTION 55
Refer to the exhibit.

A call mode through the Cisco Unified Border Element to pilot 2000 is foiling. What is causing the call to foil?

  • A. The destination pattern is incorrect for the dialed number.
  • B. No codecs are configured on the dial peers
  • C. VAD was not disabled on the outgoing dial poor.
  • D. The Cisco Unified Border Element is not receiving a response to its OPTION keepahves.

Answer: A

 

NEW QUESTION 56
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 global" configuration mode
  • B. "ephone-dn' configuration mode
  • C. "dial-peer cor custom" configuration mode
  • D. "telephony-service" configuration mode
  • E. "voice register pool" configuration mode

Answer: B,E

 

NEW QUESTION 57
A single site reports that when they dial select numbers, the call connects, but they do not get audio. The administrator finds that the calls are not routing out of the normal gateway but out of another site's gateway due to a TEHO configuration. What is the next step to diagnose and solve the issue?

  • A. Verify that IP routing is correct between the gateway and the IP phone.
  • B. Verify that the route pattern is not blocking calls to the destination number.
  • C. Verify that the dial peer of the gateway has the correct destination pattern configured.
  • D. Verify that the route pattern has the correct calling-party transformation mask

Answer: C

 

NEW QUESTION 58
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 bind media source-interface Loopback0
  • B. voice-class sip resources priority mode passthrough
  • C. voice-class sip early-offer forced.
  • D. voice-class sip bind control source-interface Loopback0

Answer: A

 

NEW QUESTION 59
Which configuration element of a hunt group allows for changing Calling Party Transformations settings?

  • A. route group
  • B. line group
  • C. hunt list
  • D. hunt pilot

Answer: D

Explanation:
Section: Cisco Unified CM Call Control Features
Explanation/Reference:
Reference: https://community.cisco.com/t5/ip-telephony-and-phones/call-alerting-on-hunt-group-as-shared-line/ td-p/2658015

 

NEW QUESTION 60
Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?

  • A. Enable the Detailed Call Analysis option under Enterprise Parameters for Unified CM.
  • B. By default, the Translation Patterns operations are printed in SDL traces, so no additional configuration is necessary.
  • C. Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to TranslationAndAlternatePatternAnalysis.
  • D. Check the Translation Patterns Analysis check box in Micro Traces on the Cisco Unified CM Serviceability page.

Answer: A

Explanation:
Reference:
https://community.cisco.com/t5/collaboration-voice-and-video/taking-sip-call-trace-on-cisco-unified- cm-using-rtmt/ta-p/3161200

 

NEW QUESTION 61
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. Enable Use Calling Party's External Phone Number Mask on the translation pattern.
  • B. Change the partition of the translation pattern from none to pstn_pt.
  • C. Enable Force Authorization Code on the route pattern.
  • D. Disable Use Calling Party's External Phone Number Mask on the route pattern.

Answer: D

 

NEW QUESTION 62
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?

  • A. The RTP traffic is arriving beyond the jitter buffer on the receiving end.
  • B. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
  • C. Cisco Unified Communications Manager invoked media termination point resources.
  • D. A firewall in the media path is blocking TCP ports 16384-32768.

Answer: D

 

NEW QUESTION 63
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