Questions for the 300-815 were updated on : Nov 29 ,2024
Topic 1
Refer to the exhibit. In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone
user
C. What are two results from this action? (Choose two.)
A C
Topic 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?
D
Topic 1
An administrator is troubleshooting a one-way audio issue for a call that uses H.323 protocol in slow-start mode. The
administrator requests that the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audio
call is provided. The H.225 and H.245 messages for one of the one-way audio calls are gathered and the call flow has not
invoked any media resources. Where is the RTP IP and port information for both sides found?
B
Explanation:
Reference: http://ccievoicehopeful.blogspot.com/2012/09/h323-notes.html
Topic 1
Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP
interworking)? (Choose two.)
A B
Topic 1
When an administrator troubleshoots H.323 call setup, which message gives an alert that the called party is being notified
about the call?
C
Topic 1
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?
C
Topic 1
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
D
Topic 1
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? (Choose two.)
B C
Topic 1
What is first preference condition matched in a SIP-enabled incoming dial peer?
A
Explanation:
Reference: https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ipvoip/211306-In-Depth-
Topic 1
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.)
A C
Topic 1
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?
C
Explanation:
Reference: https://www.cisco.com/c/en/us/support/docs/unified-communications/unifiedcommunications-manager-
callmanager/213583-procedure-to-analyse-call-flow-of-sip-ca.html
Topic 1
What is a description of RTP timestamps or sequence numbers?
D
Explanation:
Reference: https://www.cs.columbia.edu/~hgs/rtp/faq.html
Topic 1
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?
A
Topic 1
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?
C
Topic 1
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?
A