What is a function of the metadata carried in SIP sessions between the recording client and the recording server?

It identifies the participant change due to transfers during the call.

It forks RTP media to the recorder.

It provides advanced capabilities, such as speech analytics.

It identifies the participant change due to transfers during the call.

Some users report poor quality when they travel to another continent and make calls. This issue applies only to one continent and not to others, where typically the dialing is fast and quality is clear. Users experience the same result at home when they call the same numbers in that specific continent. It seems like some users do not exist in the correct PSTN gateway when making calls to a specific country. The company is using TEHO to save on the cost of international or long-distance calling.They are also using a globalized dial plan. What is the cause of the issue?

The users are missing this specific gateway at the device pool level.

CUBE is not configured forTEHO in the specific country

A local route group is not added to the route pattern.

The users are missing this specific gateway at the device pool level.

Regions in Cisco UCM are not configured correctly.

Which elements does Cisco cloud mobility for collaboration include?

Cisco Collaboration Cloud and Cisco Mobile and Remote Access Collaboration Cloud Services

Cisco Webex Collaboration Cloud Services

Cisco Collaboration Cloud and Cisco Webex Collaboration Cloud Services

Cisco Collaboration Cloud and Cisco Mobile and Remote Access Collaboration Cloud Services

Refer to the exhibit. Refer to the exhibit. A company is using Microsoft Teams with Cisco Unified Border Element integration, but the administrator sees a one-way audio issue with Microsoft Teams. The administrator must modify the SIP profile to send the proper information on the SDP for IP address for media to match the internal and external interface. Which set of commands resolves the issue?

voice class sip-profiles 1request INVITE sdp-header Connection add ''2\.76\.1'' ''2.78.1''

voice class sip-profiles 1request INVITE sdp-header Connection-Info modify ''2\.76\.1'' ''2.78.1''

voice class sip-profiles 1request INVITE sdp-header Session-Owner modify ''27\.0\.0'' ''27.3.0''

voice class sip-profiles 1request INVITE sdp-header Connection add ''2\.76\.1'' ''2.78.1''

voice class sip-profiles 1request INVITE sdp-header Audio-Attribute modify ''2\.76\.1'' ''2.71.1''

An administrator configured Extension Mobility on Cisco UCM. Users report that logins are successful, but the phones do not have an Extension Mobility option after logging in. The administrator verified that Extension Mobility is enabled on the devices and that the log-out profile is valid. Which action must the administrator take to resolve the issue?

Subscribe all device profiles to the Extension Mobility phone service.

Delete the identity trust list file from the phone(s).

Change the Extension Mobility URL from HHTP to HTTPS.

Restart the Cisco Extension Mobility and Cisco Extension Mobility application services.

An engineer has a hunt group with some overflow and many calls are going to voicemail. To reduce the number of calls forwarded to voicemail, the engineer must create Call Queuing to make callers wait before going to voicemail. How will Call Queuing affect Forward Hunt No Answer and Forward Hunt Busy?

Alter Call Queuing is enabled. Forward Hunt No Answer and Forward Hunt Busy must be disabled manually.

Forward Hunt No Answer and Forward Hunt Busy must be enabled while Call Queuing is enabled.

Alter Call Queuing is enabled. Forward Hunt No Answer and Forward Hunt Busy must be disabled manually.

Before Call Queuing is enabled. Forward Hunt No Answer and Forward Hunt Busy must be disabled manually.

When Call Queuing is enabled. Forward Hunt No Answer and Forward Hunt Busy are disabled automatically.

The company implemented Cisco Unified Mobility on the Cisco UCM. The users are satisfied and can transfer voice calls between devices. Mobile users want to extend the timer that controls the time given to pick up a voice call on an IP phone from 4 to 6 seconds. Which configuration change satisfies this requirement?

Change the ''Maximum Wait Time for Mobility Pickup'' setting from 4000 to 6000.

Change the ''Maximum Wait Time for Desk Pickup'' setting from 4 to 6

Change the ''Maximum Wait Time for Mobility Pickup'' setting from 4 to 6.

Change the ''Maximum Wait Time for Mobility Pickup'' setting from 4000 to 6000.

Change the ''Maximum Wait Time for Desk Pickup'' setting from 4000 to 6000.

Refer to the exhibit. Refer to the exhibit. An engineer is trying to set up a new deployment using the SIP provider with TCP for signaling. After troubleshooting, the customer notices that the matching incoming dial peer and outgoing dial peer did not generate the INVITE to the SIP provider. Why is the call failing?

The ITSP is not answering the SYN.

The ITSP is not sending the ACK.

The ITSP is sending CLOSE state.

The ITSP is not answering the SYN.

An administrator is configuring a SIP trunk to an ITSP. The SIP connection will traverse from a Cisco UCM to the ISTP through a Cisco Unified Border Element. The ITSP has indicated that they require an in-band method for DTMF. Which command on the outbound dial-peer to the ITSP will meet this requirement?

router (config-dial-peer) dtmf-relay rtp-nte

router (config-dial-peer) dtmf-relay sip-notify

router (config-dial-peer) dtmf-relay sip-kpml

router (config-dial-peer) dtmf-relay h245-alphanumeric

router (config-dial-peer) dtmf-relay rtp-nte

A new solution is configured to support internal, local, and international calling. Calling [+44 1111 1111] from one of the registered internal phones does not work. Local and internal calls seem to work without any problems. The configuration has patterns configured to match the failing dialed number [+44 .... ....]. The other configured patterns show [2...] for internal numbers and [555 ...] for local numbers. International numbers use E.164 as recommended. What is missing to make this solution work?

001 or 00 must be used instead of the + sign on Cisco UCM

=+ cannot be used in a route pattern, only in a SIP pattern

Cisco 300-815 Practice Test 6 - Free Online Exam Questions 201-219

Phone A calls to phone B, but phone B has Call Forward All set to a PSTN number. The route list responsible for the phone B call to the PSTN has a standard local route group configured. Which route group must be used to send the call to the PSTN?

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Cisco 300-815 Practice Test 6

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