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Cisco 300-815 Exam
Implementing Cisco Advanced Call Control and Mobility Services (CLASSM) (Page 3 )

Updated On: 12-Feb-2026
Page 3 of 50
PDF with 245 Questions

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

Answer(s): C



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 UCM 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.

Answer(s): D



An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which two debugs should the Administrator turn on? (Choose two.)

  1. H.323 messages
  2. H.225 asn1
  3. H.245 asn1
  4. H.225 media
  5. H.323 asn1

Answer(s): B,C



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






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