The Best Cisco 350-801 Study Guides and Dumps of 2021
Top Cisco 350-801 Exam Audio Study Guide! Practice Questions Edition
Understanding functional and technical aspects of Implementing Cisco Collaboration Core Technologies (350-801 CLCOR) Call Control
The following will be discussed in CISCO 350-801 dumps:
- Using Certificates
- Configure Regions and Locations
- Describe Mobile and Remote Access (MRA)
- Translation patterns
- Configure Self-Provisioning
- SIP route patterns
- Explore the Cisco VoIP Bandwidth Calculator
- Describe the Cisco Unified Communications Manager digit analysis process
- Troubleshoot Calling Issues
- Configure globalized call routing in Cisco Unified CM
- Route patterns (traditional and +E.164 format)
- Examine Cisco IOS Gateway Inbound and Outbound Dial-Peer Functions
- Transforms
- Configure IP Network Protocols
- Implement and Troubleshoot Digit Manipulation on a Cisco IOS Gateway
- Configure and Troubleshoot LDAP Integration in Cisco Unified Communications Manager
- Implement Endpoint Addressing and Call Routing
- Standard local route group
- Configure and Troubleshoot Integrated Services Digital Network (ISDN) Primary Rate Interface (PRI)
- Deploy an IP Phone Through Auto and Manual Registration
- Configure Batch Provisioning
- Implement toll fraud prevention on Cisco Unified CM
- Implement PSTN Calling Using MGCP Gateways
- Configure and Troubleshoot Collaboration Endpoints
Certification Path for Implementing Cisco Collaboration Core Technologies (350-801 CLCOR)
This course will help you:
- Implement Cisco Unified Communications Manager provisioning features
- Earn 64 credits toward recertification
- Configure and troubleshoot collaboration endpoints
- Integrate and troubleshoot Cisco Unified Communications Manager with Lightweight Directory Access Protocol (LDAP) for user synchronization and user authentication
This exam is for:
- Network administrators
- Students preparing to take the CCNP Collaboration certification
- Systems engineers
- Network engineers
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NEW QUESTION 67
Calls are being delivered to the end user in the globalized format. Where does an engineer configure the calling number into a localized format?
- A. gateway
- B. IP phone
- C. route pattern
- D. service parameters
Answer: B
NEW QUESTION 68
An engineer implements QoS in the enterprise network. Which command can to verify the correct classification and marking on a cisco IOS switch?
- A. show policy-map interface GigabitEthernet 1/0/1
- B. show policy-map
- C. show access-lists
- D. show class-map interface GigabitEthernet 1/0/1
Answer: A
NEW QUESTION 69
What is a QoS requirement for VoIP calls?
- A. ≤ 15% packet loss
- B. ≤ 150 ms of one-way latency from mouth to ear
- C. ≤ 150 bps per phone of guaranteed bandwidth
- D. ≤ 150 ms of jitter
Answer: B
Explanation:
Section: QoS
Explanation/Reference: https://www.ciscopress.com/articles/article.asp?p=357102#:~:text=One%2Dway%20latency%20 (mouth%20to,and%20Layer%202%20media%20overhead).
NEW QUESTION 70
A collaboration engineer must configure Cisco Unified Border Element to support up to five concurrent outbound calls across an Ethernet link with a bandwidth of 160 kb to the Internet Telephony Service Provider.
Which set of commands allows the engineer to complete the task without compromising voice quality?
- A. dial-peer voice 1 voip
translation-profile outgoing Strip9
max-conn 5
destination-pattern 91[2-9]..[2-9]......$
session protocol sipv2
session target ipv4:142.45.10.1
dtmf-relay rtp-nte sip-notify sip-kpml
codec aacld - B. dial-peer voice 1 voip
translation-profile outgoing Strip9
max-conn 5
destination-pattern 91[2-9]..[2-9]......$
session protocol sipv2
session target ipv4:142.45.10.1
dtmf-relay rtp-nte sip-notify sip-kpml
codec mp4a-latm - C. dial-peer voice 1 voip
translation-profile outgoing Strip9
max-conn 5
destination-pattern 91[2-9]..[2-9]......$
session protocol sipv2
session target ipv4:142.45.10.1
dtmf-relay rtp-nte sip-notify sip-kpml
codec ilbc mode 20 - D. dial-peer voice 1 voip
translation-profile outgoing Strip9
max-conn 5
destination-pattern 91[2-9]..[2-9]......$
session protocol sipv2
session target ipv4:142.45.10.1
dtmf-relay rtp-nte sip-notify sip-kpml
Answer: C
NEW QUESTION 71
Which field of a Real-Time Transport Protocol packet allows receiving devices to detect lost packets?
- A. CSRC (Contributing source ID)
- B. timestamp
- C. sequence number
- D. SSRC (Synchronization source identifier)
Answer: C
Explanation:
Section: QoS
Explanation/Reference:
https://www.oreilly.com/library/view/packet-guide-to/9781449339661/ch04.html
NEW QUESTION 72
An engineer is designing a high availability and failover solution for two Cisco Unified Border Element routers. The first router (cube 1.abc.com) takes 60% of the calls and the second router (cube2 abc.com) takes 40% of the calls. Assume all DNS A records have been created. Which two SRV records are needed for a load balanced solution? (Choose two.)
- A. _sip_udp.abc.com 60 IN SRV 3 60 cube2.abc.com
- B. _sip_udp.abc.com 60 IN SRV 1 40 cube2.abc.com
- C. _sip_udp.abc.com 60 IN SRV 60 1 cube1abc.com
- D. _sip_udp.abc.com 60 IN SRV 1 60 cube1.abc.com
- E. _sip_udp.abc.com 60 IN SRV 2 60 cube1.abc.com
Answer: B,D
NEW QUESTION 73
What is a software-based media resource that is provided by the Cisco IP Voice Media Streaming Application?
- A. auto-attendant
- B. annunciator
- C. video conference bridge
- D. transcoder
Answer: B
Explanation:
Section: Infrastructure and Design
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab09/clb09/media.html
NEW QUESTION 74
A user dials 9011841234567 to reach Vietnam. Which steps send the call to the PSTN provider as 011841234567?
A)
B)
C)
D)
- A. Option D
- B. Option B
- C. Option A
- D. Option C
Answer: C
NEW QUESTION 75
Refer to the exhibit.
The translation rule is configured on the voice gateway to translate DNIS. What is the outcome if the gateway receives 0255-343-1234 as DNIS?
- A. The translation rule is not matched because DNIS contains "-".
- B. The translation rule is matched and the translated number is 02553431234.
- C. The translation rule is not matched because DNIS does not end with a "$".
- D. The translation rule is matched and the translated number is 025553431234.
Answer: B
NEW QUESTION 76
An engineer is configuring a BOT device for a Jabber user in Cisco Unified Communication Manager Which phone type must be selected?
- A. Cisco Dual Mode for iPhone
- B. third-party SIP device
- C. Cisco Unified Client Services Framework
- D. Cisco Dual Mode for Android
Answer: D
Explanation:
Section: Collaboration Applications
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/11_5/ CJAB_BK_D00D8CBD_00_deployment-installation-guide-cisco-jabber115/ CJAB_BK_D00D8CBD_00_deployment-installation-guide-cisco-jabber115_chapter_01000.html
NEW QUESTION 77
Refer to the exhibit. What is the registration state of the analog port in this debug output?
- A. The analog port is registered to Cisco UCM.
- B. The MGCP Gateway is not communicating with the Cisco UCM.
- C. The analog port failed to register to Cisco UCM with an error code 200.
- D. The analog port is currently shut down.
Answer: A
NEW QUESTION 78
Which recommendation is the best practice for marking video and voice media in a cisco Unified Communications network?
A)
B)
C)
D)
- A. Option D
- B. Option A
- C. Option C
- D. Option B
Answer: D
NEW QUESTION 79
On a Cisco Catalyst Switch, which command is required to send CDP packets on a switch port that configures a Cisco IP phone to transmit voice traffic in 802.1Q frames, tagged with the voice VLAN ID 221?
- A. Device(config-if)# switchport vlan voice 221
- B. Device(config-if)# switchport voice vlan 221
- C. Device(config-if)# switchport access vlan 221
- D. Device(config-if)# switchport trunk allowed vlan 221
Answer: B
Explanation:
Section: Infrastructure and Design
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/switches/lan/catalyst2960/software/release/12-2_40_se/ configuration/guide/scg/swvoip.pdf
NEW QUESTION 80
Refer to the exhibit.

A call is failing to establish between two SIP Devices. The Called device answer with this SDP. Which SDP parameter causes this issue?
- A. The calling device did not offer a ptime value.
- B. The RTP port is set to 0.
- C. The payload for G.711ulaw must be 18.
- D. The media stream is set to sendonly.
Answer: B
NEW QUESTION 81
Which wildcard must an engineer configure to match a whole domain in SIP route patterns?
- A. !
- B. *
- C. .
- D. @
Answer: B
Explanation:
Explanation/Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmcfg/ CUCM_BK_C95ABA82_00_admin-guide-100/CUCM_BK_C95ABA82_00_admin-guide-
100_chapter_0100111.html
NEW QUESTION 82
What is the maximum DNS SRV entries that should be defined in the SIP Trunk destination address field in Cisco UCM?
- A. 0
- B. 1
- C. 2
- D. 3
Answer: B
NEW QUESTION 83
......
Valid 350-801 Exam Updates - 2021 Study Guide: https://www.dumps4pdf.com/350-801-valid-braindumps.html
350-801 Certification - The Ultimate Guide: https://drive.google.com/open?id=1YV0tRfY6vBr9LcVaig3X9tfkhc8d0UjT