An administrator is implementing a new dial-plan on Cisco Unified Border Element. The administrator must ensure that incoming dial-peers are matched based on the IP address from where the incoming request originates. Which dial-peer configuration should be applied to accomplish this requirement?

dial-peer voice 1 voip incoming url via

dial-peer voice 1 voip incoming url request

dial-peer voice 1 voip incoming called-number

dial-peer voice 1 voip incoming url to

An engineer must configure call queuing under a Hunt Pilot. After the engineer receives the audio file that will be played to callers during queuing, which two steps should be taken to complete the configuration? (Choose two.)

Upload the audio file in 'TFTP File Management'' via OS Administration GUI

Assign the uploaded audio file to 'Network Hold MOH Source & Announcements' under Hunt Pilot's Queuing section.

Assign the uploaded audio file to the hunting Line Group member's 'User Hold MOH Audio Source

Assign the uploaded audio file to the hunting Line Group member's 'Network Hold MOH Audio Source'.

Upload the audio file in 'TFTP File Management'' via OS Administration GUI

Assign the uploaded audio file to 'Network Hold MOH Source & Announcements' under Hunt Pilot's Queuing section.

Upload the audio file in 'MOH Audio File Management' via CM Administration GUI

Which two types of authentication are supported for the configuration of Intercluster Lookup Service? (Choose two.)

FQDN of the servers defined in DNS

Refer to the exhibit. In an active SIP call between phone user A and phone user B, phone A initiates a call transfer to phone user C. Which two scenarios are correct? (Choose two.)

Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings.

Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section.

Phone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_B User Hold MOH Audio Source settings.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the music on hold and the MOH audio is chosen from Phone_A Network Hold MOH Audio Source settings.

As soon as Phone_A presses the Transfer button for the first time, Phone_B hears the MOH and the MOH audio is chosen from Phone_A User Hold MOH Audio Source settings.

Refer to the exhibit. Users report that when they dial to Cisco Unity Connection from an external network, they cannot enter any digits. Assuming only in-band DTMF is supported, what is a reason for this malfunction?

The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work.

There is SIP Delayed Offer. DTMF is supported only in Early Offer.

The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.

End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?

Contact: header of the 200 OK response

Allow: header if the 200 OK response

Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?

The RTP traffic is arriving beyond the jitter buffer on the receiving end.

The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.

Cisco Unified Communications Manager invoked media termination point resources.

The RTP traffic is arriving beyond the jitter buffer on the receiving end.

A firewall in the media path is blocking TCP ports 16384-32768.

Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. All calls are established, and sometimes they work well, but sometimes there is one-way audio or no audio. You determine that there is a firewall between the floors, and the administrator reports that it is allowing SIP signaling and UDP ports from 20000 to 22000 bidirectionally. What are two possible solutions? (Choose two.)

Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767

Ask the firewall administrator to change the range of UDP ports to 16384-32767.

Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767

Ask the firewall administrator to change the ports to TCP.

Ask the firewall administrator to change the range of UDP ports to 16384-32767.

Go to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.

Go to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.

Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?

Voice/Video > Session Trace Log View > Real Time Data

Analysis Manager > Inventory > Trace File Repositories

System > Tools > Trace and Log Central

Voice/Video > Session Trace Log View > Real Time Data

Voice/Video > Session Trace Log View > Open From Local Disk

Which description of RTP timestamps or sequence numbers is true?

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

The sequence number is used to detect losses.

Timestamps increase by the time ''carrying'' by a packet.

Sequence numbers increase by four for each RTP packet transmitted.

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

Refer to the exhibit. A user reports that when they call a specific phone number, no one answers the call, but when they call from a mobile phone, the call is answered. The engineer troubleshooting the issue is expecting the far-end gateway to cut through audio on the 183 Session Progress SIP message. Which SIP Profile configuration element is necessary for the Cisco Unified Communications Manager to send acknowledgement of provisional responses?

On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.

Allow Passthrough of Configured Line Device Caller Information must be enabled.

Accept Audio Codec Preferences in Received Offer must be set to On.

On the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages.

Early Offer for G Clear Calls must be enabled.

Which action is correct with respect to toll fraud prevention configuration in the Cisco Unified Communications Manager Express?

Configure IP Address Trusted Authentication for Incoming VoIP Calls.

Configure Direct Inward Dial for Incoming ISDN Calls with overlap dialing.

Configure IP Address Trusted Authentication for Incoming VoIP Calls.

Configure the command no ip address trusted authenticate under ''voice service voip''.

Enable Secondary Dial tone on Analog and Digital FXO Ports.

You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration. Which configuration is occurring in this section?

configuration for a single SIP phone

configuration items common for all SIP phones

configuration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)

configuration for SIP registrar service

For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?

interworking between an OOB method and RFC2833 for flow-around calls

interworking between h245-signal and rtp-nte

interworking between an OOB method and RFC2833 for flow-through calls

interworking between h245-alpha numeric and sip-kpml

Where is the dtmf-relay command configured on Cisco Unified Border Element?

in the voice-class VoIP configuration

Refer to the exhibit. Calls incoming from the provider are not working through newly set up Cisco Unified Border Element. Provider engineers get the 404 Not Found SIP message. Incoming calls are coming from the provider with called number ''222333444'' and Cisco Unified Communications Manager is expecting the called number to be delivered as ''444333222''. The administrator already verified that the IP address of the Cisco Unified CM is set up correctly and there are no dial peers configured other than those shown in the exhibit. Which action must the administrator take to fix the issue?

Set up translation-profile on the incoming dial peer to match incoming traffic.

Change the destination-pattern on the outgoing dial peer to match ''444333222''.

Set up translation-profile on the incoming dial peer to match incoming traffic.

Create specific matching for ''222333444'' on the incoming dial peer.

Fix the voice translation-rule to match specifically number ''222333444'' and change it to ''444333222''.

Refer to the exhibit. Users report that outbound PSTN calls from phones registered to Cisco Unified Communications Manager are not completing. The local service provider in North America has a requirement to receive calls in 10-digit format. The Cisco Unified CM sends the calls to the Cisco Unified Border Element router in a globalized E.164 format. There is an outbound dial peer on Cisco Unified Border Element configured to send the calls to the provider. The dial peer has a voice translation profile applied in the correct direction but an incorrect voice translation rule applied, which is shown in the exhibit. Which rule modified DNIS in the format that the provider is expecting?

rule 1/^\+1$[2-9]..[2-9]......$$/ /\1/

rule 1 /^$[2-9]..[2-9]......$$/ /\1/

rule 1 /^\+1$[2-9]..[2-9]......$$/ /\0/

Refer to the exhibit. An administrator is troubleshooting why users are not hearing audio when dialing long distance numbers across their Cisco Unified Border Element. The customer's carrier has a requirement that dialing long distance requires an access code to be entered. Looking at the exhibit, what two actions can be taken to correct signaling? (Choose two.)

Enable Early Offer on the Cisco Unified Border Element.

Enable the supplementary-service media-renegotiate command.

Enable Mid-Call Signaling Consumption.

An engineer must route all SIP calls in the form of <user>@example.com to the SIP trunk gateway corporate local. Which two SIP route patterns can be used to accomplish this task? (Choose two.)

example.com@gateway.corporate.local

Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration? (Choose two.)

The Client Matter Code feature does not support overlap sending because the Cisco Unified CM cannot determine when to prompt the user for the code.

If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.

The Client Matter Code feature does not support overlap sending because the Cisco Unified CM cannot determine when to prompt the user for the code.

If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.

If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.

The Client Matter Code feature does support overlap sending because the Cisco Unified Communications Manager can determine when to prompt the user for the code.

The Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization Code.

Which two descriptions of the Standard Local Route Group deployment are true? (Choose two.)

can be associated only under the route list

chooses the route group that is configured under the device pool of the calling-party device

can be associated under the route group

can be associated only under the route list

chooses the route group that is configured under the device pool of the calling-party device

chooses the route group that is configured under the device pool of the called-party device

can be assigned directly to the route pattern

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?

incoming called number 555.......

After configuring a Cisco CallManager Express with Cisco Unity Express, inbound calls from the PSTN SIP trunk receive a ring tone for 20 seconds and then a busy signal instead of voicemail. Which configuration fixes this problem?

Router(config)# voice service voip Router(conf-voi-serv)#no supplementary-service sip moved-temporarily

Router(config)# voice service voip Router(conf-voi-serv)#allow-connections h323 to h323

Router(config)#dial-peer voice 2 voip Router(config-dial-peer)#no vad

Router(config)# voice service voip Router(conf-voi-serv)#allow-connections voice-mail mod

Router(config)# voice service voip Router(conf-voi-serv)#no supplementary-service sip moved-temporarily

An engineer must configure a secure SIP trunk with a remote provider, with a specific requirement to use port5065 for inbound and otubound traffic. Which two items must be configured to complete this configuration? (Choose two.)

Destination Port in SIP Information section of the SIP Trunk configuration

Incoming Port in SIP Trunk Security Profile configuration

Incoming Port in SIP Information section of the SIP Trunk configuration.

Incoming Port in Security Information of the SIP Profile configuration.

Destination Port in SIP Information section of the SIP Trunk configuration

Incoming Port in SIP Trunk Security Profile configuration

Destination Port in SIP Trunk Security Profile configuration

Cisco 300-815 Practice Test 1 - Free Online Exam Questions 1-40

Question 1 / 40

An administrator is configuring a cluster for ILS and wants to limit the amount of entities that Cisco Unified Communications Manager can write to the database for data that is learned through ILS. Which service parameter is used to adjust this limit?

ILS Max Number of Learned Objects in Database

ILS Active Learned Object Upper Limit

Global Data Service Parameter Limit

Imported Dial Plan Replication Database Object Lower Limit

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