Cisco CCNP 300-815 Questions & Answers

• Question 171:

  

  Refer to the exhibit. The customer is troubleshooting an issue where users cannot dial the CMS SIP Route Pattern. The CMS URI they are attempting to dial is [email protected]. Which IPv4 pattern must the customer enter to resolve the issue?

  A. [email protected]

  B. *@cms.domain.local

  C. *.cms.domain.local

  D. cms.domain.local

  Correct Answer: B

• Question 172:

  

  Refer to the exhibit. An administrator just implemented SIP trunking on their Cisco UCM and reports that calls using the SIP trunk are using Media Termination Point resources unnecessarily. Which action resolves the issue?

  A. Change DTMF Signaling Method to “No Preference”.

  B. Disable SIP Rel1XX Options.

  C. Change to a range that does not result in MTP.

  D. Change DTMF Signaling Method to “RFC 4733”.

  Correct Answer: C

• Question 173:

  An engineer is configuring a Cisco UCM solution. The requirements state that some users in one location will receive calls from a number during work hours 09AM to 5PM, and another group will get the calls from the same number outside this defined timeslot. Users will also change the outgoing number when reaching out to customers based on the same time-of-day routing rules. Which feature is needed to allow for this type of configuration?

  A. partitions

  B. CSS

  C. route groups

  D. route lists

  Correct Answer: B

• Question 174:

  A company is experiencing an issue where calls from the Cisco UCM via Cisco Unified Border Element to the ITSP fail with a “500 server internal error” right after a 183 session in progress is sent back from the ITSP. What must be enabled to resolve the issue?

  A. delayed offer on Cisco UCM

  B. early offer at the CUBE

  C. delayed offer on CUBE

  D. early offer at the ITSP

  Correct Answer: B

• Question 175:

  

  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.

  Correct Answer: B

• Question 176:

  An SRST site must use pickup functionality on the Chicago remote site to allow the users to take incoming calls. The pickup command is configured under the call manager-fallback section on the SRST router. What are two results of that configuration? (Choose two.)

  A. Calls that ring an unassigned directory number are forwarded to the auto-attendant.

  B. The PickUp soft key is on all SRST phones.

  C. The GPickUp soft key is on all SRST phones.

  D. Calls that come into one directory number can be picked up from another directory number.

  E. Calls that come into a Cisco UCM registered phone can be picked up by a SRST phone.

  Correct Answer: BD

• Question 177:

  

  Refer to the exhibit. Users report that outgoing calls do not work on the new SIP trunk for outgoing calls. The solution consists of a Cisco UCM Cluster linked to a Cisco Unified Border Element where the SIP trunk is terminated. The provider required 10 digits. The logs show a line going toward the Cisco Unified Border Element. Which code snippet must be added to the configuration to meet the requirement?

  A. request Invite sip-header modify “andamp;lt;sip:1(…)@” “andamp;lt;sip:9135551\1@” under the SIP translation profile configuration

  B. request Invite sip-header modify “andamp;lt;sip:1(…)@” “andamp;lt;sip:9135551\1@” under the voice translation profile configuration

  C. sip-header modify “andamp;lt;sip:1(…)@” “andamp;lt;sip:9135551\1@” under the voice translation profile configuration

  D. request Invite sip-header Diversion modify “andamp;lt;sip:1(…)@” “andamp;lt;sip:9135551\1@” under the SIP profile configuration

  Correct Answer: B

• Question 178:

  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

  Correct Answer: B

• Question 179:

  A customer reports audio quality issues between video endpoints in the HQ location in California and one of the branches in Texas. Which two actions must RTCP take to troubleshoot this issue? (Choose two.)

  A. Allow for VAD to be used for calls using the G7.29 codec, which reduces the usage of the WAN bandwidth and saves around 30% of bandwidth per call

  B. Configure the rtcp keepalive command in Cisco Unified Border Element to generate reports, which can be reviewed using the debug voip rtcp packet command.

  C. Encrypt the media to stop rogue devices from replying and putting that traffic back on the WAN, which avoids any extra bandwidth and ensures the quality of the calls.

  D. Gather statistics on a media connection and information such as packets sent, lost packets, jitter, feedback, and round-trip delay. This information can help isolate the type of audio quality issues and the direction of the affected traffic.

  E. Compress the headers of RTP traffic to lower the bandwidth consumption over the WAN, which allows more calls with less bandwidth consumed.

  Correct Answer: BD