Pass the Cisco CCNP Collaboration 300-815 Questions and answers with CertsForce

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Questions # 21:

Refer to the exhibit.

Question # 21

An administrator is troubleshooting a problem in which some outbound calls from an internal network to the Internet telephony service provider are not getting connected, but some others connect successfully. The firewall team found that some call attempts on port 5060 came from an unrecognized IP that has not been defined in the firewall rule. What should the administrator configure in the Cisco Unified Border Element to fix this issue?

Options:

A.

use of port 5061 for SIP secure


B.

access list allowing the firewall IP


C.

ip prefix-list to filter the unwanted IP address


D.

bind signaling and media to the loopback interface


Expert Solution
Questions # 22:

The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real- Time Transport Protocol traffic that had the one-way audio call.

You gather the H.225 and H.245 messages for one of the one-way audio calls. Where can you find the RTP IP and port information for both sides? (Note: This call flow has not invoked any media resources like MTP or transcoders).

Options:

A.

H.245 Terminal Capability Set


B.

H.245 Open Logical Channel


C.

H.225 Connect


D.

H.245 Open Logical Channel Ack


Expert Solution
Questions # 23:

Question # 23

Refer to the exhibit. A Collaboration engineer is implementing call routing and must configure the Cisco Unified Border Element to register to multiple registrars using a SIP trunk listen port.

Which code snippet completes the configuration?

Options:

A.

session transport tcp


B.

session transport tls


C.

dtmf-relay rtp-nte


D.

bind control source-interface GigabitEthernet0/0


Expert Solution
Questions # 24:

Which IOS command creates a SIP-enabled dial peer?

Options:

A.

voice dial-peer 20 sip


B.

dial-peer voice 20 voip


C.

dial-peer voice 20 pots


D.

dial peer voice 20 sip


Expert Solution
Questions # 25:

Refer to the exhibit.

Question # 25

A collaboration engineer is troubleshooting an issue where the PSTN calls of a Cisco UCM IP phone user are not reaching the PSTN gateway. Which action resolves the issue?

Options:

A.

Deselect "Block this pattern" on the route pattern.


B.

Change the calling search space of The user's line or device.


C.

Change the "Call Classification" to "OnNet" on the route pattern.


D.

Ensure that the user's phone is assigned to a device pool with the correct local route group settings.


Expert Solution
Questions # 26:

What are two functions of the metadata in SIP recording? (Choose two.)

Options:

A.

includes the details of the application media stream


B.

contains the cryptographic context of the SRTP passthrough calls


C.

identifies the session and media association time


D.

carries the Agent Statistics Report


E.

carries the communication session data (audio and video calls) that describes the call to the recording server


Expert Solution
Questions # 27:

In Cisco Unified Communications Manager, which tool do you use to check SIP traces?

Options:

A.

MTP


B.

CCSIP


C.

RTMT


D.

OS Administration Page


Expert Solution
Questions # 28:

An engineer set up and successfully tested a TEHO solution on the Cisco UCM. PSTN calls are routed correctly using the IP WAN as close to the final PSTN destination as possible. However, suddenly, calls start using the backup local gateway instead. What is causing the issue?

Options:

A.

WAN connectivity


B.

LAN connectivity


C.

route pattern


D.

route list and route group


Expert Solution
Questions # 29:

Which description of RTP timestamps or sequence numbers is true?

Options:

A.

The sequence number is used to detect losses.


B.

Timestamps increase by the time “carrying” by a packet.


C.

Sequence numbers increase by four for each RTP packet transmitted.


D.

The timestamp is used to place the incoming audio and video packets in the correct timing order (playout

delay compensation).


Expert Solution
Questions # 30:

When a third-party SIP Phone System is dialed inbound across a Cisco Unified Border Element, DTMF is failing. The third-party vendor accepts only out-of-band DTMF. Which configuration should be added to the outgoing dial peer to resolve this issue?

Options:

A.

dtmf-relay h245-signal


B.

dtmf-relay rtp-nte


C.

dtmf-relay cisco-rtp


D.

dtmf-relay sip-kpml


Expert Solution
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