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Download Implementing Cisco Advanced Call Control and Mobility Services Exam Dumps
NEW QUESTION 49
In Cisco Unified Communications Manager globalized call routing is implemented and must confirm that it is correctly implemented without making a call. Which tool do you use for verification?
- A. Dialed Number Analyzer
- B. SDI trace
- C. SDL trace
- D. Real-Time Monitoring Tool
Answer: A
Explanation:
Section: Call Control and Dial Planning
NEW QUESTION 50
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?
- A. Analysis Manager > Inventory > Trace File Repositories
- B. Voice/Video > Session Trace Log View > Open From Local Disk
- C. System > Tools > Trace and Log Central
- D. Voice/Video > Session Trace Log View > Real Time Data
Answer: D
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
NEW QUESTION 51
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. c= line of SDP content
- B. Allow: header if the 200 OK response
- C. o= line of SDP content
- D. Contact: header of the 200 OK response
Answer: A
NEW QUESTION 52
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. incoming called number 555.......
- B. answer-address 555 ........
- C. session-protocol sipv2
- D. codec g729
Answer: C
NEW QUESTION 53
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. incoming called number 555.......
- B. answer-address 555 ........
- C. session-protocol sipv2
- D. codec g729
Answer: C
NEW QUESTION 54
......
