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Free download Cisco certification 300-815 exam practice questions and answers

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Cisco 300-815 Exam covers a wide range of topics, including call control protocols, Cisco Unified Communications Manager, Cisco Unity Connection, Cisco Unified IM and Presence, Cisco Expressway, and more. It is designed to test candidates' ability to implement and troubleshoot these technologies in a real-world environment, as well as their ability to design and deploy advanced call control and mobility services.

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Cisco 300-815 Exam covers a wide range of topics related to advanced call control and mobility services. Candidates will be tested on their ability to configure and troubleshoot CUCM features such as call routing, multi-site deployment, media resources, and SIP trunks. They will also be evaluated on their knowledge of Cisco Unity Connection features such as voicemail, unified messaging, and AutoAttendant. Additionally, the exam covers Cisco Unified IM&P features such as presence, chat, and group chat.

Cisco Implementing Cisco Advanced Call Control and Mobility Services Sample Questions (Q208-Q213):

NEW QUESTION # 208
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.)

A. transcoding
B. T.38 fax relay
C. SIP trunk
D. secure SIP lines
E. three-way conference

Answer: B,E

Explanation:
The following supplementary services are supported as part of IPv6 in Unified SRST IP Phones:
Hold/Resume
Call Forward
Call Transfer
Three-way Conference (with BIB conferencing only)
Line to T1/E1 Trunk and Trunk to Line with Supplementary Service Features Fax to and from PSTN (IPv4 ATA to ISDN T1/E1) for both T.38 Fax Relay and Fax Passthrough
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/ guide/SCCP_and_SIP_SRST_Admin_Guide/ srst_sip_isr4000.html

NEW QUESTION # 209
A customer reports audio quality issues between video endpoints in the HQ location in California and one of the branches in Texas. Which two actions must RTCP take to troubleshoot this issue? (Choose two.)

A. Encrypt the media to stop rogue devices from replying and putting that traffic back on the WAN, which avoids any extra bandwidth and ensures the quality of the calls.
B. Allow for VAD to be used for calls using the G7.29 codec, which reduces the usage of the WAN bandwidth and saves around 30% of bandwidth per call.
C. Configure the rtcp keepalive command in Cisco Unified Border Element to generate reports, which can be reviewed using the debug voip rtcp packet command.
D. Compress the headers of RTP traffic to lower the bandwidth consumption over the WAN. which allows more calls with less bandwidth consumed.
E. Gather statistics on a media connection and information such as packets sent, lost packets, jitter, feedback, and round-trip delay. This information can help isolate the type of audio quality issues and the direction of the affected traffic.

Answer: D,E

NEW QUESTION # 210
Refer to the exhibit. A Cisco Unified Border Element continues to send 180/183 with the required:
100rel header to Cisco UCM. and the call eventually disconnects
How is the issue resolved?

A. Disable "Send send-receive SDP in mid-call INVITE* on the SIP Profile Configuration Page in Cisco UCM.
B. Enable *Early Offer support for voice and video calls" on the SIP Profile Configuration Page in Cisco UCM.
C. Enable 'SIP ReI1XX Options* and -Early Offer Support" on the SIP Profile Configuration Page in Cisco UCM.
D. Disable "SIP Rel1XX Options* and 'Early Offer Support* on the SIP Profile Configuration Page in Cisco UCM.

Answer: B

NEW QUESTION # 211
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. You determine 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 possible solutions? (Choose two.)

A. Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.
B. Ask the firewall administrator to change the range of UDP ports to 16384-32767.
C. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767
D. Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
E. Ask the firewall administrator to change the ports to TCP.

Answer: B,D

Explanation:
In this case, one solution is to change the ports on the SIP profile of the phones to match the ports on the firewall. The other solution is to ask the firewall admin to change his ports to match the ports in UCM.
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/port/9_1_1/CUCM_BK_T2CA6EDE
_00_tcp-port-usage-guide-91/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-
91_chapter_01.html

NEW QUESTION # 212
Users are reporting that several inter-site calls are failing, and the message "not enough bandwidth" is showing on the display. Voice traffic between locations goes through corporate WAN. and Call Admission Control is enabled to limit the number of calls between sites.
How is the issue solved without increasing bandwidth utilization on the WAN links?

A. Disable Call Admission Control and let the calls use the amount of bandwidth they require.
B. Configure Call Queuing so that the user waits until there is bandwidth available
C. Reroute all calls through the PSTN and avoid using WAN.
D. Configure AAR to reroute calls that are denied by Call Admission Control through the PSTN.

Answer: D

NEW QUESTION # 213
......

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