Cisco 300-815 Practice Test - Questions Answers, Page 16
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DRAG DROP
An organization configures a SIP trunk in Cisco UCM to connect to another system. These requirements must be met:
1. Use a specific IP address for SIP signaling.
2. Encrypt the signaling traffic.
3. Restrict which devices can use the SIP trunk. 4 Simplify SIP signaling.
Drag and drop the Cisco UCM configuration steps from the left onto the order on the right to achieve these goals.
Users in a small location report that audio calls work Tine, but only one video call works before the next one gets a busy signal. Three video conference rooms were recently installed by a third party. The audio and video devices are Cisco endpoints. The site uses video and audio calls, and a network investigation does not show saturation on the link The site is configured with QoS and CoS. All devices are registered on the headquarters Cisco UCM duster. What is the cause of this problem?
The subzone setting for video is too low.
The zone settings for video are too low.
The location setting for video is too low.
The pipe or link configuration for video is set to only one device.
Refer to the exhibit.
An administrator just Implemented SIP trunking on their Cisco UCM and reports that calls using the SIP trunk are using Media Termination Point resources unnecessarily. Which action resolves the issue?
Disable SIP Red XX Options.
Change to a range that does not result in MTP.
Change OTMF Signaling Method to 'No Preference'.
Change DTMF Signaling Method to 'RFC 4733'.
Refer to the exhibit.
Refer to the exhibit. An engineer deploys CAC to Cisco UCM. UptofiveG.711 calls must be supported on the WAN link between Site A and Site B. Users report that only four concurrent calls are possible between Site A and Site B. Why isn't the filth concurrent call successful?
The G.729 audio codec is negotiated between Site A and B instead of the G.711 codec.
The QoS configuration on the WAN link causes the fifth call to be dropped.
Cisco UCM is using alternative links because of the Weight value.
The Audio Bandwidth value is undersized and must be set to 400 kbps.
Refer to the exhibit.
Refer to the exhibit. Calls from users to the PSTN in an organization get disconnected with a 408 Request Timeout when the called party is unavailable to pick up the call. Which solution must be used to resolve this challenge?
Configure midcall-signaling preserve-codec.
Choose 'Send PRACK' if 1xx contains SDP in the SIP profile.
Configure midcall-signaling passthru media-change
Choose 'Send PRACK for all 1xx Messages' in the SIP profile.
Alter a new SIP trunk is configured from a third-party SIP trunk provider, the call does not seem to work. A troubles hooting tool and a tiace log from the Cisco Unified Border Element shows that there is communication with the provider and that it is sending out SDP information in the 2XX response. Why is the call not working?
The Cisco Unified Border Element does not have any PDVM resources to terminate the SIP trunk assigned.
The SIP provider expects a delayed offer.
The SIP provider expects an early offer.
The Cisco Unified Border Element must be licensed for third-party SIP trunks.
An SRST site must use pickup functionality on the Chicago remote site to allow the users to take incoming calls. The pickup command is configured under the call-manager-fallback section on the SRST router What are two results of that configuration? (Choose two.)
The PickUp soft key is on all SRST phones.
The GPickUp soft key is on all SRST phones.
Calls that come into one directory number can be picked up from another directory number.
Calls that come into a Cisco UCM registered phone can be picked up by a SRST phone.
Calls that ring an unassigned directory number are forwarded to the auto-attendant.
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?
Call Routing > Route/Hunt > Call Park
Call Routing > Route/Hunt > Hunt Pilot
Call Routing > Route/Hunt > Line Group
Call Routing > Route/Hunt > Hunt List
Refer to the exhibit.
Refer to the exhibit An engineer deploys a Cisco Unified Border Element for SIP trunk interconnection to an ITSP The engineer performs test calls but there is no audio when the called party picks up the phone. What must the engineer do to resolve the issue?
Disable SIP User Agent bind status (media).
Decrease the SIP max-forwards value to 16.
Disable SIP early-media for 180 responses with SDP
Enable Check media source packets.
Refer to the exhibit.
An administrator just upgraded to Cisco UCM to version 14 and started SIP implementation with some new SIP trunks. During the testing, an error was reported when making a call. Which action resolves the issue?
Change to a valid range.
Change DTMF Signaling Method to 'RFC 4833*.
Make sure the destination number is configured correctly and the device is registered.
Disable SIP Rel1XX Options.
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