Ensure a flawless pass rate with the current 350-801 dumps

Embark on your certification odyssey, with the unwavering compass of the 350-801 dumps guiding you. Precisely aligned to the intricate tapestry of the curriculum, the 350-801 dumps present a wide swath of practice questions, forging an indomitable mastery. Whether you gravitate towards the structured coherence of PDFs or the dynamic scenarios painted by the VCE format, the 350-801 dumps cater adeptly. A detailed study guide, a cornerstone of the 350-801 dumps, illuminates the path, spotlighting crucial touchpoints. Asserting our staunch belief in the quality of these resources, we proudly put forth our 100% Pass Guarantee.

Lay the groundwork for 350-801 triumph with our complimentary VCE resources, updated with recent questions

Question 1:

An engineer is integrating Unity Connection with Cisco UCM. Which two actions must be configured so that recording and playback from the IP phones works at all times, including peak traffic hours? (Choose two.)

A. Increase the number of voice ports.

B. If it\’s a Unity Connection Cluster, ensure that replication is fine and not in split-brain mode.

C. The phone system to which the phones are registered in Unity Connection has the Default Trap Switch check box enabled.

D. Add dedicated dial-out ports with the allow trap connections setting selected.

E. Ensure that you have set up SIP Digest Authentication on the SIP trunk security profile.

Correct Answer: AD



Question 2:

According to QoS guidelines, what is the packet loss for streaming video?

A. Not more than 8%

B. Not more than 1%

C. Not more than 3%

D. Not more than 5%

Correct Answer: D

Streaming-Video

When addressing the QoS needs of Streaming-Video traffic, the following guidelines are recommended:

Streaming-Video (whether unicast or multicast) should be marked to DSCP CS4, as designated by the QoS Baseline.

Loss should be no more than 5 percent.

https://www.ciscopress.com/articles/article.asp?p=357102andseqNum=2



Question 3:

Refer to the exhibit.

When making a call to a MRA client, what are the combinations of protocol on each of the different sections A-B-C?

A. IP TCP/TLS(A) +SIP TCP/TLS (B) +TLS (C)

B. SIP TLS (A) +SIP TLS (B) +SIP TLS (C)

C. SIP TCP/TLS (A) + SIP TLS (B) + SIP TLS (C)

D. SIP TCP/TLS +SIP TCP/TLS (B) + SIP TCP/TLS (C)

Correct Answer: C

https://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/expressway/config_guide/X12-5/Cisco-Expressway-IP-Port-Usage-for-Firewall-Traversal-Deployment-Guide-X12-5.pdf



Question 4:

What is the maximum DNS SRV entries that should be defined in the SIP Trunk destination address field in Cisco UCM?

A. 4

B. 8

C. 1

D. 16

Correct Answer: C



Question 5:

An engineer implements a new Cisco UCM based telephony system per these requirements:

1.

The local Ethernet bandwidth is sized based on the total bandwidth per call.

2.

A G.736 codec is used.

3.

The bit rate is 64 kbps.

4.

The codec sample interval is 10 ms.

5.

The voice payload size is 160 bytes per 20 ms.

What should the size of the Ethernet bandwidth be per call?

A. 31.2 kbps

B. 38.4 kbps

C. 55.2 kbps

D. 87.2 kbps

Correct Answer: D

To calculate the Ethernet bandwidth per call, we need to take into account the total number of bytes per second in each direction (transmit and receive) and add additional overhead for Ethernet, IP, and UDP headers.

The total number of bytes per second is calculated as follows:

160 bytes per 20 ms = (160 bytes/20 ms) x (50 packets/s) = 8000 bytes/s

The bit rate is 64 kbps, so we need to add an additional 8 kbps for overhead:

64 kbps + 8 kbps = 72 kbps

The total number of bytes per second including overhead is:

72,000 bps / 8 bits per byte = 9,000 bytes/s

Adding additional overhead for Ethernet, IP, and UDP headers, we can estimate that the total number of bytes per second will be approximately 12,000 bytes/s in each direction. Therefore, the Ethernet bandwidth per call should be:

12,000 bytes/s x 8 bits per byte = 96 kbps

Therefore, the correct answer is D. 87.2 kbps is not sufficient to support the required bandwidth per call.



Question 6:

A user reports transfer failure from an IP phone for calls received from a PSTN to another PSTN number. What is a reason for these failures?

A. The IP phone is configured with the wrong region.

B. The incoming calling search space of the SIP trunk does not include the partition of the line in the IP phone.

C. The service parameter related to Offnet to Offnet Call Transfer is set to TRUE.

D. The gateway is configured with the wrong device pool.

Correct Answer: C



Question 7:

A network administrator with ID392116981 has determined that a WAN link between two Cisco UCM clusters supports only 1 Mbps of bandwidth for voice traffic How many calls does this link support if G711 as the audio codec is used?

A. 15

B. 16

C. 13

D. 12

Correct Answer: D



Question 8:

What is a valid class included in the 8-Class QoS Strategy in a VoIP network?

A. Assured Forwarding

B. Broadcast Video

C. Multimedia Conferencing

D. Real-Time Interactive

Correct Answer: C

Reference: https://www.ciscopress.com/articles/article.asp?p=2756478andseqNum=8



Question 9:

An engineer configures a SIP trunk for MWI between a Cisco UCM cluster and Cisco Unity Connection. The Cisco UCM cluster fails to receive the SIP notify messages. Which two SIP trunk settings resolve this issue? (Choose two.)

A. transmit security status

B. accept unsolicited notification

C. allow charging header

D. accept out-of-band notification

E. accept out-of-dialog refer

Correct Answer: BE



Question 10:

Which two parameters influence the total number of supported conference participants on a Cisco IOS XE router that has DSP modules? (Choose two.)

A. voice codec

B. session capacity of the PVDM module

C. number of protocol data units

D. software version Of the router

E. license types

Correct Answer: AB

These are two parameters that influence the total number of supported conference participants on a Cisco IOS XE router that has DSP modules. A voice codec is a software algorithm that compresses and decompresses voice signals for transmission over a network. Different voice codecs have different bandwidth requirements and quality levels. A PVDM module is a hardware component that provides digital signal processing (DSP) resources for voice applications such as conferencing and transcoding. A PVDM module has a fixed session capacity, which is the maximum number of voice channels that it can support simultaneously.



Question 11:

What does average rate limiting allow?

A. transmits traffic bursts up to the Bc size

B. more traffic than the CIR to be sent when there is available bandwidth

C. bandwidth up to the Be size

D. traffic to burst to the Be size when there is available bandwidth

Correct Answer: D

Reference: https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/qos_plcshp/configuration/15-mt/qos-plcshp-15-mt-book/qos-plcshp-oview.html



Question 12:

How does an administrator make a Cisco IP phone display the last 10 digits of the calling number when the call is in the connected state, and also display the calling number in the E.164 format within call history on the phone?

A. Configure a translation pattern that has a Calling Party Transform Mask of XXXXXXXXXX.

B. On the inbound SIP trunk, change Significant Digits to 10.

C. Change the service parameter Apply Transformations On Remote Number to True.

D. Configure a calling party transformation pattern that keeps only the last 10 digits.

Correct Answer: D

Translationpattern might work, but have to be triggered as called number equals directory number, not suitable sip-trung can change digits on calling number Calling Party Transformations on IP Phones Calling Party Transformation Patterns allow the system to adapt the calling party numbers to different formats. The most typical use is to adapt from globalized to localized calling party numbers and vice versa. https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab11/collab11/dialplan.html



Question 13:

Refer to the exhibit.

A call to an international number has failed. Which action corrects this problem?

A. Assign a transcoder to the MRGL of the gateway.

B. Strip the leading 011 from the called party number

C. Add the bearer-cap speech command to the voice port.

D. Add the isdn switch-type primart-dms100 command to the serial interface.

Correct Answer: B



Question 14:

A company hosts a conference call with no local users. How does the administrator stop the conference from continuing?

A. remove the transcoder

B. modifies the Block OffNet to OffNet Transfer service parameter

C. changes the codecs that are supported on the conference resource

D. modifies the Drop Ad Hoc Conference service parameter

Correct Answer: D



Question 15:

Which two protocols should be configured for the cisco unity connection and cisco UCM integration?

A. SIP

B. H.323

C. MGCP

D. RTP

E. SCCP

Correct Answer: AE

Reference: https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/11x/design/guide/b_11xcucdg/b_11xcucdg_chapter_00.html#ID-2342-00000133


Leave a Reply

Your email address will not be published. Required fields are marked *