Download Implementing Cisco Advanced Call Control and Mobility Services (CLASSM).300-815.BrainDumps.2020-04-21.36q.vcex

Vendor: Cisco
Exam Code: 300-815
Exam Name: Implementing Cisco Advanced Call Control and Mobility Services (CLASSM)
Date: Apr 21, 2020
File Size: 1 MB

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Demo Questions

Question 1
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?
  1. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.
  2. There is SIP Delayed Offer. DTMF is supported only in Early Offer.
  3. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
  4. No DTMF is negotiated.
Correct answer: D
Question 2
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?
Correct answer: C
Question 3
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?
  1. Contact: header of the 200 OK response
  2. Allow: header if the 200 OK response
  3. o= line of SDP content
  4. c= line of SDP content
Correct answer: C
Question 4
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?
  1. debug H.323 messages
  2. debug H.225 asn1
  3. debug H.246 asn 1
  4. debug H.225 media
  5. debug H.323 asn 1
Correct answer: B
Question 5
What is first preference condition matched in a SIP-enabled incoming dial peer?
  1. incoming uri
  2. target carrier-id
  3. answer-address
  4. incoming called-number
Correct answer: A
Question 6
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
Question 7
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?
  1. CallManager traces
  2. CTI Manager traces
  3. Cisco IP Manager Assistant
  4. Call logs
Correct answer: A
Question 8
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?
  1. Allow Passthrough of Configured Line Device Caller Information must be enabled.
  2. Accept Audio Codec Preferences in Received Offer must be set to On.
  3. On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.
  4. Early Offer for G Clear Calls must be enabled.
Correct answer: C
Question 9
A company has an SRST gateway running an IOS XE image. The company plans to enable the IPv6 addressing companywide. To enable the IPv6 in a unified SRST gateway to support SIP phones, what are two supported supplementary features for an IPv6 fallback scenario? (Choose two.)
  1. three-way conference
  2. secure SIP lines
  3. T.38 fax relay
  4. transcoding
  5. SIP trunk
Correct answer: AC
Question 10
Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?
  1. Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.
  2. Configure IP Address Trusted Authentication for Incoming VoIP Calls.
  3. Configure the command no ip address trusted authenticate under “voice service voip”.
  4. Enable Secondary Dial tone on Analog and Digital FXO Ports.
Correct answer: B

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