Download Implementing Cisco Advanced Call Control and Mobility Services (CLASSM).300-815.DumpsBase.2021-05-04.78q.vcex

Vendor: Cisco
Exam Code: 300-815
Exam Name: Implementing Cisco Advanced Call Control and Mobility Services (CLASSM)
Date: May 04, 2021
File Size: 3 MB
Download Exam

How to open VCEX files?

Files with VCEX extension can be opened by ProfExam Simulator.

Download
ProfExam Simulator

Purchase
Coupon: EXAM_HUB

Discount: 20%

ProfExam Simulator ProfExam Simulator ProfExam Simulator ProfExam Simulator ProfExam Simulator

Demo Questions

Question 1
Refer to the exhibit. 
   
  
The Cisco Unified Border Element receives an INVITE matching inbound dial peer 5002. The outbound dial peer supports only iLBC. and a Local Transcoding Interface is allocated.Based on the configuration and SDP from the INVITE message, which codec is chosen by Cisco Unified Border Element for the inbound call leg?
  1. G.711 A-law
  2. G.711 U-law
  3. G.729r8
  4. G.729br8
Correct answer: C
Question 2
Which configuration must an administrator perform to display Translation Pattern operations in Cisco Unified Communications Manager SDL traces?
  1. Enable the Detailed Call Analysis option under Enterprise Parameters for Unified CM.
  2. Set up the Digit Analysis Complexity in Service Parameters for Cisco Unified CM to Translation And Alternate Pattern Analysis.
  3. Check the Translation Patterns Analysis check box in Micro Traces on the Cisco Unified CM Serviceability page.
  4. By default, the Translation Patterns operations are printed in SDL traces, so no additional configuration is necessary.
Correct answer: D
Explanation:
Reference: https://community.cisco.com/t5/collaboration-voice-and-video/taking-sip-call-trace-on-cisco-unified-cm-using-rtmt/ta-p/3161200  
Reference: 
https://community.cisco.com/t5/collaboration-voice-and-video/taking-sip-call-trace-on-cisco-unified-cm-using-rtmt/ta-p/3161200  
Question 3
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?
  1. Analysis Manager > Inventory > Trace File Repositories
  2. System > Tools > Trace and Log Central
  3. Voice/Video > Session Trace Log View > Real Time Data
  4. Voice/Video > Session Trace Log View > Open From Local Disk
Correct 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
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 4
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
  1. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
  2. Cisco Unified Communications Manager invoked media termination point resources.
  3. The RTP traffic is arriving beyond the jitter buffer on the receiving end.
  4. A firewall in the media path is blocking TCP ports 16384-32768.
Correct answer: D
Question 5
Refer to the exhibit. 
   
  
An administrator is troubleshooting a problem in which some outbound calls from an internal network to the Internet telephony service provider are not getting connected, but some others connect successfully. The firewall team found that some call attempts on port 5060 came from an unrecognized IP that has not been defined in the firewall rule. What should the administrator configure in the Cisco Unified Border Element to fix this issue?
  1. use of port 5061 for SIP secure
  2. access list allowing the firewall IP
  3. ip prefix-list to filter the unwanted IP address
  4. bind signaling and media to the loopback interface
Correct answer: D
Question 6
Refer to the exhibit. 
   
  
A user dials 84969010 and observes that the call is not routed immediately. The administrator notices that after matching the fixed-length translation pattern, the call hits the \+! pattern and waits for interdigit timeout. What should be configured to ensure that the call routes out immediately?
  1. Allow Device Override on the route pattern
  2. Route Next Hop By Calling Party Number on the translation pattern
  3. Do Not Wait For Interdigit Timeout On Subsequent Hops on the translation pattern
  4. Do Not Wait For Interdigit Timeout On Subsequent Hops on the route pattern
Correct answer: C
Question 7
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?
  1. R2(config-ephone-dn)#park reservation-group 60
  2. R2(config-ephone-dn)#park-slot timeout 60 limit 2 recall alternate 3002
  3. R2(config-ephone-dn)#park reservation-group 1
  4. R2(config-ephone-dn)#park-slot timeout 30 limit 2 recall alternate 3002
Correct answer: D
Question 8
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?
  1. call-blast
  2. parallel
  3. sequential
  4. peer
Correct answer: D
Question 9
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?
  1. Router(config)# voice service voip 
    Router(conf-voi-serv)#allow-connections h323 to h323
  2. Router(config)#dial-peer voice 2 voip 
    Router(config-dial-peer)#no vad
  3. Router(config)# voice service voip 
    Router(conf-voi-serv)#allow-connections voice-mail mod
  4. Router(config)# voice service voip 
    Router(conf-voi-serv)#no supplementary-service sip moved-temporarily
Correct answer: D
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_call_handling.html
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_call_handling.html
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.)
  1. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
  2. Ask the firewall administrator to change the ports to TCP.
  3. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
  4. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
  5. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
Correct 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  
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  
CONNECT US
HOW TO OPEN VCE FILES

Use VCE Exam Simulator to open VCE files
Avanaset

HOW TO OPEN VCEX AND EXAM FILES

Use ProfExam Simulator to open VCEX and EXAM files
ProfExam Screen

ProfExam
ProfExam at a 20% markdown

You have the opportunity to purchase ProfExam at a 20% reduced price

Get Now!