An engineer is troubleshooting local ringback on a Cisco SIP gateway The gateway is not ignoring the SIP 180 response with SDP from the service provider, and the far end device is sending the 180 with SDP to play ringback from the IP address specified m the SDP Which configuration change must be made on the gateway to resolve the issue?

Router(conftg-sip-ua)# disable-early-media 180

Router(conf-voi-serv)# dlisable-early-media 180

Router(conftg-sip-ua)# disable-early-media 180

Router(con(-voi-serv)# no disable-early-media 180

Router(config-sip-ua)# no disable-early-media 180

An engineer has temporarily disabled toll fraud prevention for SIP line calls on a Cisco CME12.6x and must enforce security and toll fraud prevention for the SIP line side on Cisco Unified CME. Which configuration must be used to start this process?

voice service volp ip address trusted authenticate

voice service volp Ip address trusted list

voice service volp enablo ip address trust authentication

voice service volp enable Ip address trust list

voice service volp ip address trusted authenticate

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?

Configure the service parameter hunt group logoff notification and specify the name of the ringtone file.

Configure the HLog softkey on the phone so that while a user is logged off, it plays an audible tone when a call is missed.

Set the service parameter Party Entrance Tone to True.'

Configure the service parameter hunt group logoff notification and specify the name of the ringtone file.

Set the service parameter Enterprise Feature Access number for hunt group logout and set up an access number

An engineer needs to deploy Cisco Extension Mobility. The engineer already activated the required services, configured the Extension Mobility Phone Service, and created an Extension Mobility device profile that is associated with end users. Which two additional steps are required to complete the configuration? (Choose two.)

Subscribe the device profile to the Extension Mobility Phone Service.

Define the default Device Template on IP phones when users log out.

Subscribe IP phones to the Extension Mobility Phone Service.

Consolidate Cisco UCM certificates by using Bulk Certificate Management.

Subscribe directory numbers to the Extension Mobility Phone Service.

Subscribe the device profile to the Extension Mobility Phone Service.

Define the default Device Template on IP phones when users log out.

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?

Mark the \+ [2-9JXX[2-91XXXXXX route pattern with urgent priority.

Change the Cisco UCM routing logic to the longest-match algorithm.

Select Allow Overlap Sending for the \+! route pattern.

Mark the \+ [2-9JXX[2-91XXXXXX route pattern with urgent priority.

Set the interdigit timeout service parameter value to 0.

An organization needs to ensure that the Cisco UCM can provide a resume softkey to users on their desk phones when they disconnect an anchored call on their mobile phone. Which solution must be used to accomplish these goals?

Set the 'Retain Media on Disconnect with PI for Active Call' service parameter to False.

Set the 'Retain Media on Disconnect with PI for Active Cal' service parameter to True

Refer to the exhibit. An administrator received reports that users do not hear a ringback when they dial specific external numbers. For example, the Cisco Unified Border Element receives a delayed offer invite from Cisco UCM and responds with the 183 session progress in the exhibit. Assume that the PSTN is responsible for generating ringback in these problematic scenarios. Which action resolves the issue?

Edit the configuration for SIP ReMXX Options in the appropriate SIP profile on Cisco UCM to Send PRACK if 1xx Contains SDP.

Enable early media on the appropriate SIP trunk on Cisco UCM.

Edit the configuration for SIP ReMXX Options in the appropriate SIP profile on Cisco UCM to Send PRACK if 1xx Contains SDP.

On Cisco Unified Border Element: voice service voip sip Re1xx require 100rel

On Cisco Unified Border Element: voice service voip sip Re1xx disable 100rel

An engineer needs to deploy the local route group feature to a Cisco UCM server by using the standard local route group. For each device pool, the engineer needs to specify a different PSTN access for emergency calls. What must the engineer do?

Add a new standard local route group. Select a route group as the new standard local route group in Device Pools. Create a route pattern of 911 that points to a route list that contains the new standard local route group.

Create a new local route group name. In Device Pools, select a route group for that new local route group. Create a route pattern of 911 that points to a route list that contains the new local route group.

Set a new local route group name. Select a route group as the new local route group in Device Pools. Create a route pattern of 911 that points to the new local route group.

Add a new standard local route group. Select a route group as the new standard local route group in Device Pools. Create a route pattern of 911 that points to the new standard local route group.

What is a capability of the gateway-preferred, network-based recording option in Cisco UCM?

Cisco UCM integrates with recording destination via SIP trunk.

It requires SIP trunk and CTI with recording application.

Cisco UCM integrates with recording destination via SIP trunk.

It uses BIB of IP phone to fork audio to the recording destination.

It initiates when the application (recorder) dictates that it must be initiated.

Refer to the exhibit. A collaboration engineer is troubleshooting an issue where external callers cannot leave voicemail messages. Also, internal users report hearing the reorder tone (fast busy) when they attempt to retrieve voicemail messages from their Cisco IP phones. Which action resolves the issue?

Ensure that Cisco UCM can resolve the destination address via DNS.

Start the Cisco Call Manager service at the destination.

Ensure that the SIP Trunk Security Profile is configured to use UDP for transport.

Verify that the correct port numbers are used for the SIP trunk.

The company Cisco UCM cluster has two different gateways for off-net calls. The current configuration uses 9 as a prefix to get to the main gateway. The secondary gateway is for any calls that start with 9713, but this is not yet configured. The admin does not want to add more route patterns other than the current 9 prefix for the gateway to the Cisco UCM. How must the Cisco UCM be configured to meet the requirements?

Configure two CSSs. then add a translation pattern with the secondary CSS to send calls with 9713 to the secondary gateway.

Configure two partitions and two CSSs, then add a translation pattern with the secondary CSS to send all calls with 9713 to the secondary gateway.

Create a Standard Route Group to dynamically route calls with prefix 9713 to the secondary gateway.

Configure a Route Group and a Route List to send calls with prefix 9713 to the secondary gateway.

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?

Increase the bandwidth allowance between the RTP_Reg and SJ_Reg regions to 64 kbps.

Disable G.722 on all devices at both sites.

Enable Early Offer on the SIP trunk.

Increase the bandwidth allowance between the RTP_Reg and SJ_Reg regions to 64 kbps.

Create a new audio codec preference list with G.711 U-law 64k as the highest priority and apply it to RTP_Reg and SJ_Reg.

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 Desk Pickup' setting from 4000 to 6000.

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

Change the 'Maximum Wait Time for Desk 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.

An engineer needs to configure a translation pattern in Cisco UCM to match dialed numbers that start with 9011 so that the calls can be routed with a single route pattern of VH If the engineer uses translation pattern 9011.! which two called party transformation settings must be configured in Cisco UCM? (Choose two.)

set Called Party Transform Mask to +!

set Prefix Digits (Outgoing Calls) to +

set Called Party Transform Mask to +!

set Called Party Numbering Plan to International

set Called Party Number Type to International

A customer reports audio quality issues between video endpoints in the HQ location in California and one of the branches in Texas. Which two actions must RTCP take to troubleshoot this issue? (Choose two.)

Compress the headers of RTP traffic to lower the bandwidth consumption over the WAN. which allows more calls with less bandwidth consumed.

Gather statistics on a media connection and information such as packets sent, lost packets, jitter, feedback, and round-trip delay. This information can help isolate the type of audio quality issues and the direction of the affected traffic.

Compress the headers of RTP traffic to lower the bandwidth consumption over the WAN. which allows more calls with less bandwidth consumed.

Configure the rtcp keepalive command in Cisco Unified Border Element to generate reports, which can be reviewed using the debug voip rtcp packet command.

Allow for VAD to be used for calls using the G7.29 codec, which reduces the usage of the WAN bandwidth and saves around 30% of bandwidth per call.

Encrypt the media to stop rogue devices from replying and putting that traffic back on the WAN, which avoids any extra bandwidth and ensures the quality of the calls.

Gather statistics on a media connection and information such as packets sent, lost packets, jitter, feedback, and round-trip delay. This information can help isolate the type of audio quality issues and the direction of the affected traffic.

A collaboration engineer is configuring call pickup for a Cisco UCM user that currently has a working BLF speed dial button for the teammate's extension. The user wants to see calls ringing on the teammate's extension and to answer those calls by pressing the BLF speed dial button. Which configuration step meets this requirement?

Assign the extensions to different call pickup groups.

Set the call pickup group to <none> for both extensions.

Select the call pickup checkbox on the BLF speed dial settings.

Assign both extensions to the same partition.

Assign the extensions to different call pickup groups.

A customer reports that outbound calls from Cisco UCM to Cisco Unified Border Element are not getting early media. During troubleshooting, the customer finds that Cisco UCM is sending a delayed offer SIP INVITE. Cisco Unified Border Element responds with a 183 Session Progress with SDP message. However, the customer finds that Cisco UCM does not respond to the 183 messages. Which action must the customer perform to resolve the problem?

On Cisco Unified Border Element, configure early-offer forced.

On Cisco Unified Border Element, configure midcall-signaling passthru.

On Cisco UCM, change the 'Early Offer support for voice and video calls' field on the SIP profile that is applied to the SIP trunk to Disabled.

On Cisco Unified Border Element, configure early-offer forced.

On Cisco UCM. change the 'SIP ReHXX Options' field on the SIP profile that is applied to the SIP trunk to Send PRACK if 1xx Contains SDP.

A Collaboration engineer is implementing call processing capabilities on the Cisco UCM and needs to configure Call Queuing to place callers in a queue until hunt members are available to answer them. Which action must be completed before this feature can be configured?

Service the Cisco IP Voice Media Streaming (IPVMS) application.

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

contains the cryptographic context of the SRTP passthrough calls

carries the Agent Statistics Report

includes the details of the application media stream

contains the cryptographic context of the SRTP passthrough calls

identifies the session and media association time

carries the Agent Statistics Report

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

Refer to the exhibit. A collaboration engineer investigates an issue where the user calls that match the local Jabber domain are routed back to Cisco UCM. Which action resolves this problem?

Create another route SIP pattern for the local domain with the block pattern selected.

Check the check box to block the pattern for the local domain.

Create another route SIP pattern for the local domain with the block pattern selected.

Create a translation pattern to block calls to the local domain.

Create a transformation pattern to block calls to the local domain.

An engineer configures several calling party transformation patterns in Cisco UCM so that calling numbers are shown in the +E.164 format. The route patterns match seven-digit internal numbers and prefix the numbers with 1408. The SIP trunk matches 1408XXXXXXX and prefixes the number with +. When users make outbound calls, the calling number still shows as seven digits. What is the cause of this problem?

The engineer must also apply a calling transformation at the route group details level.

The calling transformation on the SIP trunk is applied first, and then the calling transformation on the route patterns is applied.

Calling transformations cannot be configured on both route patterns and a SIP trunk.

The calling transformation on the route patterns is ignored because there is already a calling transformation on the SIP trunk.

The engineer must also apply a calling transformation at the route group details level.

A company is using Cisco Jabber on-premises to make B2B calls on video. The calls are using Cisco Expressway-C and Expressway-E and have been configured in Cisco UCM to be able to call any URI on the internet. The Jabber client also has voice enabled and must be able to call local, regional, and international numbers. Where must Cisco UCM be configured to meet this requirement for URI dialing?

Enter '.*' in the route pattern section tied to a route group and list.

Enter '!#' in the route pattern section tied to a route group and list.

Enter '*' in the SIP route pattern.

Enter '!#' in the SIP route pattern.

Enter '.*' in the route pattern section tied to a route group and list.

Refer to the exhibit. Refer to the exhibit. Users report that outgoing calls do not work on the new SIP trunk for outgoing calls. The solution consists of a Cisco UCM Cluster linked to a Cisco Unified Border Element where the SIP trunk is terminated. The provider required 10 digits. The logs show a line going toward the Cisco Unified Border Element. Which code snippet must be added to the configuration to meet the requirement?

request Invite sip-header Diversion modify '<sip:1(...)@' '<sip:9135551\1@' under the SIP profile configuration

request Invite sip-header modify '<sip:1(...)@' '<sip:9135551\1@' under the SIP translation profile configuration

sip-header modify '<sip:1(...)@' '<sip:9135551\1@' under the voice translation profile configuration

request Invite sip-header Diversion modify '<sip:1(...)@' '<sip:9135551\1@' under the SIP profile configuration

request Invite sip-header modify '<sip:1(...)@' '<sip:9135551\1@' under the voice translation profile configuration

What does VoIP trace in a Cisco Unified Border Element do?

It stores all can trace data in system memory (or successful and failed calls

It stores all call trace data in system memory for failed calls.

It stores all can trace data in system memory (or successful and failed calls

It stores all call trace data in system log files for successful and failed calls.

It stores all can trace data in system file logs for failed calls.

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 pipe or link configuration for video is set to only one device.

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?

Change OTMF Signaling Method to 'No Preference'.

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 passthru media-change

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 SIP provider expects a delayed offer.

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.

Calls that come into one directory number can be picked up from another directory number.

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 > Hunt Pilot

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?

Decrease the SIP max-forwards value to 16.

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?

Make sure the destination number is configured correctly and the device is registered.

Change DTMF Signaling Method to 'RFC 4833*.

Make sure the destination number is configured correctly and the device is registered.

Cisco 300-815 Practice Test 4 - Free Online Exam Questions 121-160

An administrator is working on an issue between the customer s Cisco Unified Border Element and the service provider. The provider only wants to see mid-call signaling from the Cisco Unified Border Element for fax calls. Which command must be configured on Cisco Unified Border Element?

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Cisco 300-815 Practice Test 4

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