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Cisco Implementing Advanced Call Control and Mobility Services (CLASSM) 300-815 Exam Questions in PDF

Free Cisco 300-815 Dumps Questions (page: 2)

300-815 PDF — 245 Questions

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

Answer(s): A


Reference:

https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth- Explanation-of-Cisco-IOS-and-IO.html#anc8



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. It is determined 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 solutions for this issue? (Choose two.)

  1. Go to the SIP profile assigned to these IP phones in Cisco UCM 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 UCM and change the range of media ports to 20000-22000.
  5. Go to System Parameters in Cisco UCM and change the range of media ports to 20000-22000.

Answer(s): A,C



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

Answer(s): C


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



What is a description of RTP timestamps or sequence numbers?

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

Answer(s): D


Reference:

https://www.cs.columbia.edu/~hgs/rtp/faq.html



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

Answer(s): A





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.

Answer(s): C





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 provide, 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?

  1. Require: 100rel
  2. Content-Type: application/sdp
  3. Contact: <sip:987654321@192.168.100.200:5060>
  4. Content-Disposition: session;handling=required

Answer(s): A



The SIP session refresh timer allows the RTP session to stay active during an active call. The Cisco UCM sends either SIP-INVITE or SIP-UPDATE messages in a regular interval of time throughout the active duration of the call. During a troubleshooting session, the engineer finds that the Cisco UCM is sending SIP-UPDATE as the SIP session refresher, and the engineer would like to use SIP-INVITE as the session refresher.
What configuration should be made in the Cisco UCM to achieve this?

  1. Change Session Refresh Method on the SIP profile to INVITE.
  2. Increase Retry INVITE to 20 seconds on the SIP profile.
  3. Enable Send send-receive SDP in mid-call INVITE on the SIP profile.
  4. Enable SIP Rel1XX Options on the SIP profile.

Answer(s): A


Reference:

https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-border- element/213843-troubleshoot-session-refresh-on-cube.html



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