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

Refer to the exhibit.

Users report high delays when calling national numbers in the United States. An engineer analyzes the route patterns in Cisco UCM. What must be done to resolve this issue?

A customer wants to configure their team of 10 agents to answer incoming calls to a common central number. The phone for each team member must ring in sequence and ring four timesbefore moving to the next member's phone. Drag and drop the steps from the left into the order on the right to make this happen.

Exhibit.

Refer to the exhibit. All calls from site A to site B are failing, and the issue has been identified as a media negotiation problem. Which configuration change resolves the issue?

Refer to the exhibit. A collaboration engineer is troubleshooting an issue where a user of a Cisco UCM IP phone reports failed calls when trying to dial out to the PSTN. Which action resolves the issue?

A company has users that are logged in to hunt groups. However, there is a requirement for hunt group configurations to provide an option to turn on audible ringtones when calls to a line group arrive at a phone that is logged out and on a break. This ringtone alerts a logged-out user that there is an incoming call to a hunt list to which the line is a member, but the call does not ring at the phone of that line group member because of the logged-out status. Which action meets this requirement?

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

Configure Call Queuing in Cisco Unified Communications Manager. Where do you set the maximum number of callers in the queue?

An administrator wants to use the Call Queuemg feature on Cisco UCM to allow new customers to wait in the queue while other agents are not available to take their call. Which configuration step enables this feature?

Users of a Cisco UCME report that calls lose audio after they are forwarded to some users. Calling point-to-point seems to function fine, but depending on the type of phone, some calls lose audio when forwarded. Looking at the trace header that shows:

Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER

What is missing and is a reason that the SRTP stream stops working?

Refer to the exhibit.

An engineer configures Cisco Unified Border Element to connect the enterprise VoIP network with a SIP telephony provider. Calls are not working in either direction. What must be configured in the dial peer 1 to fix the issue?