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Free Cisco 300-815 CLACCM Practice Questions 2026 Part3

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Question #1
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.)
A. hone_A sends a SIP-REFER message to the Cisco Unified Communications Manager with Phone_C information in the Refer-To section
B. hone_B sends a SIP-REFER message to the Cisco Unified CM with Phone_C information in the Refer-To section
C. s 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
D. s 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
E. s 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
View answer
Correct Answer: AD

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Question #2
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?
A. Change DTMF Signaling Method to "No Preference"
B. Disable SIP Rel1XX Options
C. Change to a range that does not result in MTP
D. Change DTMF Signaling Method to "RFC 4733"
View answer
Correct Answer: D
Question #3
Refer to the exhibit. An administrator has configured a SIP trunk between two Cisco UCM clusters. For calls that should use the trunk, the calls fail with a fast busy. The administrator checks the Cisco CallManager SDL traces and found that the cluster to which the calling device is registered never sends an INVITE to the destination cluster. The administrator also verifies that all nodes from both clusters are powered on, and the CallManager service is running. How is this issue resolved?
A. The administrator must associate the route pattern with a calling search space the device can dial
B. The administrator needs to enable OPTIONS pings on the SIP trunks for both clusters
C. The administrator must allow connectivity so that TCP connections do not fail between the nodes
D. The administrator needs to disable OPTIONS pings on the SIP trunks for both clusters
View answer
Correct Answer: C
Question #4
A company has an SRST gateway running an IOS XE image. The company plans to enable the IPv6addressing companywide. To enable the IPv6 in a unified SRST gateway to support SIP phones, whatare two supported supplementary features for an IPv6 fallback scenario? (Choose two.)
A. hree-way conference
B. ecure SIP lines
C.
D. ranscoding
E. IP trunk
View answer
Correct Answer: AC
Question #5
An engineer must configure a Cisco UCM hunt list so that calls to users in a line group are routed to the first idle user and then the next.Which distribution algorithm must be configured to accomplish this task?
A. broadcast
B. top down
C. longest idle time
D. circular
View answer
Correct Answer: C
Question #6
Which action is correct with respect to toll fraud prevention configuration in the Cisco UnifiedCommunications Manager Express?
A. onfigure Direct Inward Dial for Incoming ISDN Calls with overlap dialing
B. onfigure IP Address Trusted Authentication for Incoming VoIP Calls
C. onfigure the command no ip address trusted authenticate under “voice service voip”
D. nable Secondary Dial tone on Analog and Digital FXO Ports
View answer
Correct Answer: B
Question #7
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?
A. Make sure the destination number is configured correctly and the device is registered
B. Disable SIP Rel1XX Options
C. Change DTMF Signaling Method to "RFC 4833"
D. Change to a valid range
View answer
Correct Answer: A
Question #8
Where is the dtmf-relay command configured on Cisco Unified Border Element?
A. n the voice-class VoIP configuration
B. n the VoIP dial peer
C. n global SIP configuration
D. n the VoIP or POTS dial peers
View answer
Correct Answer: B
Question #9
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?
A. Configure midcall-signaling preserve-codec
B. Choose "Send PRACK" if 1xx contains SDP in the SIP profile
C. Configure midcall-signaling passthru media-change
D. Choose "Send PRACK for all 1xx Messages" in the SIP profile
View answer
Correct Answer: D
Question #10
Some users report having issues dialing some external numbers when traveling to other locations within the company. The company has five locations in five cities in one country and has an egress gateway in each location for TEHO. The configuration has no specific entry stating that the roaming users are using the local gateway, but calls are going out. How is a verification of the call routing in such a specific configuration performed to further identify the problem?
A. device mobility
B. standard local route group
C. local route groups
D. TEHO
View answer
Correct Answer: A
Question #11
What is a function of the metadata carried in SIP sessions between the recording client and the recording server?
A. It forks RTP media to the recorder
B. It provides advanced capabilities, such as speech analytics
C. It sets up a new SIP session
D. It identifies the participant change due to transfers during the call
View answer
Correct Answer: D
Question #12
Users report that calls to some locations do not work. The phone is calling, the recipient picks up, but it does not connect. These users are in the headquarters and registered to the company Cisco UCM with default values for calling (G.711). The problem locations are smaller remote locations with low bandwidth capacity, so the phones are configured with ILBC. Which command should be used to troubleshoot the issue?
A. show dspfarm dsp active
B. show dsp active
C. debug voice dspfarm
D. debug dsp all
View answer
Correct Answer: C
Question #13
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start mode). The administrator requests thatyou provide the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audio call.You gather the H.225 and H.245 messages for one of the one-way audio calls. Where can you find the RTP IP and port information for both sides? (Note: This callflow has not invoked any media resources like MTP or transcoders).
A. H
B. H
C. H
D. H
View answer
Correct Answer: B
Question #14
What is first preference condition matched in a SIP-enabled incoming dial peer?
A. incoming uri
B. target carrier-id
C. answer-address
D. incoming called-number
View answer
Correct Answer: A
Question #15
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.)
A. Allow for VAD to be used for calls using the G7
B. Configure the rtcp keepalive command in Cisco Unified Border Element to generate reports, which can be reviewed using the debug voip rtcp packet command
C. 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
D. Gather statistics on a media connection and information such as packets sent, lost packets, jitter, feedback, and round-trip delay
E. Compress the headers of RTP traffic to lower the bandwidth consumption over the WAN, which allows more calls with less bandwidth consumed
View answer
Correct Answer: BD
Question #16
A customer has multisite deployments with a globalized dial plan. The customer wants to route PSTN calls via the gateway assigned to each site.Which two actions will fulfill the requirement? (Choose two.)
A. Create one global route list for PSTN calls that points to one global PSTN route group
B. Create a route group which has all the gateways and associate it to the device pool of every site
C. Assign one route group as a local route group in the device pool of the corresponding site
D. Create one route group for each site and one global route list for PSTN calls that point to the local route group
E. Create a hunt group and assign it to each side route pattern
View answer
Correct Answer: AC
Question #17
A customer has multisite deployments with a globalized dial plan. The customer wants to route PSTN calls via the gateway assigned to each site.Which two actions will fulfill the requirement? (Choose two.)
A. Create one global route list for PSTN calls that points to one global PSTN route group
B. Create a route group which has all the gateways and associate it to the device pool of every site
C. Assign one route group as a local route group in the device pool of the corresponding site
D. Create one route group for each site and one global route list for PSTN calls that point to the local route group
E. Create a hunt group and assign it to each side route pattern
View answer
Correct Answer: AC
Question #18
A customer has multisite deployments with a globalized dial plan. The customer wants to route PSTN calls via the gateway assigned to each site.Which two actions will fulfill the requirement? (Choose two.)
A. Create one global route list for PSTN calls that points to one global PSTN route group
B. Create a route group which has all the gateways and associate it to the device pool of every site
C. Assign one route group as a local route group in the device pool of the corresponding site
D. Create one route group for each site and one global route list for PSTN calls that point to the local route group
E. Create a hunt group and assign it to each side route pattern
View answer
Correct Answer: AC
Question #19
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?
A. Subscribe all device profiles to the Extension Mobility phone service
B. Delete the identity trust list file from the phone(s)
C. Change the Extension Mobility URL from HHTP to HTTPS
D. Restart the Cisco Extension Mobility and Cisco Extension Mobility application services
E. Reveal Answer
View answer
Correct Answer: A
Question #20
The administrator of ABC company is troubleshooting a one-way audio issue for a call that uses H.323 protocol (slow-start mode). The administrator requests that you provide the IP and port information of the Real-Time Transport Protocol traffic that had the one-way audio call.You gather the H.225 and H.245 messages for one of the one-way audio calls. Where can you find the RTP IP and port information for both sides? (Note: This call flow has not invoked any media resources like MTP or transcoders).
A.
B.
C.
D.
View answer
Correct Answer: B
Question #21
A customer has multisite deployments with a globalized dial plan. The customer wants to route PSTN calls via the gateway assigned to each site.Which two actions will fulfill the requirement? (Choose two.)
A. Create one global route list for PSTN calls that points to one global PSTN route group
B. Create a route group which has all the gateways and associate it to the device pool of every site
C. Assign one route group as a local route group in the device pool of the corresponding site
D. Create one route group for each site and one global route list for PSTN calls that point to the local route group
E. Create a hunt group and assign it to each side route pattern
View answer
Correct Answer: AC
Question #22
A company was looking at the IT charges and saw many long-distance and international calls primarily to sites in North America and around the world. The administrator wants to optimize the PSTN expense. Which dial plan configuration reduces PSTN connectivity charges by using the IP network to bring the egress point to the PSTN as close as possible to the called number?
A. Atranslation patterns
B. Btail end hop off
C. Cclient matter codes
D. Ddial rules
View answer
Correct Answer: B
Question #23
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?
A. Configure Hunt Pilot Queuing
B. Service the Cisco IP Voice Media Streaming (IPVMS) application
C. Configure Music On Hold
D. Configure announcements
View answer
Correct Answer: B
Question #24
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?
A. Contact: header of the 200 OK response
B. Allow: header if the 200 OK response
C. o= line of SDP content
D. c= line of SDP content
View answer
Correct Answer: C
Question #25
An engineer is troubleshooting Cisco Device Mobility and find that the phone has roamed to a building that is assigned to a different device pool but has not changed its device pool accordingly What action resolves the issue?
A. Set correct Location under Current Device Mobility Settings
B. Enable SRST under Current Device Mobility Settings
C. Set the correct subnet under Device Mobility Info
D. Set Device CSS under Current Device Mobility Settings
E. Reveal Answer
View answer
Correct Answer: C
Question #26
A company has an SRST gateway running an IOS XE image. The company plans to enable the IPv6 addressing companywide. To enable the IPv6 in a unified SRST gateway to support SIP phones, what are two supported supplementary features for an IPv6 fallback scenario? (Choose two.)
A. three-way conference
B. secure SIP lines
C. T
D. transcoding
E. SIP trunk
View answer
Correct Answer: AC
Question #27
A customer has multisite deployments with a globalized dial plan. The customer wants to route PSTN calls via the gateway assigned to each site.Which two actions will fulfill the requirement? (Choose two.)
A. Create one global route list for PSTN calls that points to one global PSTN route group
B. Create a route group which has all the gateways and associate it to the device pool of every site
C. Assign one route group as a local route group in the device pool of the corresponding site
D. Create one route group for each site and one global route list for PSTN calls that point to the local route group
E. Create a hunt group and assign it to each side route pattern
View answer
Correct Answer: AC
Question #28
A new deployment is using MVA for a specific user on the sales team, but the user is having issues when dialing DTMF. Which DTMF method must be configured in resolve the issue?
A. Agateway
B. Bout-of-band
C. Cchannel
D. Din-band
View answer
Correct Answer: B
Question #29
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?
A. The negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work
B. There is SIP Delayed Offer
C. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive
D. No DTMF is negotiated
View answer
Correct Answer: D
Question #30
An engineer must configure a Cisco UCM hunt list so that calls to users in a line group are routed to the first idle user and then the next.Which distribution algorithm must be configured to accomplish this task?
A. broadcast
B. top down
C. longest idle time
D. circular
View answer
Correct Answer: C
Question #31
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?
A. voice service voip enable ip address trust list
B. voice service voip ip address trusted list
C. voice service voip ip address trusted authenticate
D. voice service voip enable ip address trust authentication
View answer
Correct Answer: B
Question #32
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources forDTMF?
A. nterworking between an OOB method and RFC2833 for flow-around calls
B. nterworking between h245-signal and rtp-nte
C. nterworking between an OOB method and RFC2833 for flow-through calls
D. nterworking between h245-alpha numeric and sip-kpml
View answer
Correct Answer: A
Question #33
When you troubleshoot H.323 call setup, which message informs you that the called party is being notified about the call?
A. ALERTING
B. PROCEEDING
C. CONNECT
D. RINGING
View answer
Correct Answer: A
Question #34
You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration. Which configuration is occurring in this section?
A. onfiguration for a single SIP phone
B. onfiguration for SIP registrar service
C. onfiguration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)
D. onfiguration items common for all SIP phones
View answer
Correct Answer: A
Question #35
An administrator is configuring an Intercluster Lookup Service between 10 Cisco UCM clusters. Due to security requirements, certificate-based authentication must be used. Due to the deployment size, the administrator wants to avoid manually exchanging certificates between the clusters. Which two steps must be followed to meet these requirements? (Choose two)
A. Set all dusters to standalone on the ILS configuration page
B. Install multiserver CA-signed Tomcat certificates on all cluster publishers
C. Enable password-based authentication in conjunction with certificate-based authentication
D. Ensure that the cluster ID for all clusters is the same
E. Install multiserver CA-signed CallManager certificates on all cluster publishers
F. Reveal Answer
View answer
Correct Answer: BC
Question #36
An engineer must configure a Cisco UCM hunt list so that calls to users in a line group are routed to the first idle user and then the next.Which distribution algorithm must be configured to accomplish this task?
A. broadcast
B. top down
C. longest idle time
D. circular
View answer
Correct Answer: C
Question #37
An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which two debugs should the Administrator turn on? (Choose two.)
A. H
B. H
C. H
D. H
E. H
View answer
Correct Answer: BC
Question #38
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. What are two results from this action? (Choose two.)
A. Phone_A sends a SIP-REFER message to the Cisco UCM with Phone_C information in the Refer-To section
B. Phone_B sends a SIP-REFER message to the Cisco UCM with Phone_C information in the Refer-To section
C. 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
D. 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
E. 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
View answer
Correct Answer: AC
Question #39
An engineer must configure a Cisco UCM hunt list so that calls to users in a line group are routed to the first idle user and then the next.Which distribution algorithm must be configured to accomplish this task?
A. broadcast
B. top down
C. longest idle time
D. circular
View answer
Correct Answer: C
Question #40
Refer to the exhibit.A user reports that when they call a specific phone number, no one answers the call, but when theycall from a mobile phone, the call is answered. The engineer troubleshooting the issue is expectingthe far-end gateway to cut through audio on the 183 Session Progress SIP message. Which SIP Profileconfiguration element is necessary for the Cisco Unified Communications Manager to sendacknowledgement of provisional responses?
A. llow Passthrough of Configured Line Device Caller Information must be enabled
B. ccept Audio Codec Preferences in Received Offer must be set to On
C. n the SIP Profile, the configuration parameter SIP Rel1XX Options must be set to Send PRACK for all 1xx Messages
D. arly Offer for G Clear Calls must be enabled
View answer
Correct Answer: C
Question #41
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?
A. voice service voip enable ip address trust list
B. voice service voip ip address trusted list
C. voice service voip ip address trusted authenticate
D. voice service voip enable ip address trust authentication
View answer
Correct Answer: B
Question #42
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 in the SDP.Which configuration change must be made on the gateway to resolve the issue?
A. Router(config-sip-ua)# no disable-early-media 180
B. Router(conf-voi-serv)# no disable-early-media 180
C. Router(conf-voi-serv)# disable-early-media 180
D. Router(config-sip-ua)# disable-early-media 180
View answer
Correct Answer: D
Question #43
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.)
A. The PickUp soft key is on all SRST phones
B. The GPickUp soft key is on all SRST phones
C. Calls that come into one directory number can be picked up from another directory number
D. Calls that come into a Cisco UCM registered phone can be picked up by a SRST phone
E. Calls that ring an unassigned directory number are forwarded to the auto-attendant
View answer
Correct Answer: AC
Question #44
Which two extended capabilities must be configured on dial peers for fast start-to-early media scenarios (H.323 to SIP interworking)? (Choose two.)
A. DTMF
B. BFCP
C. VIDEO
D. FAX
E. AUDIO
View answer
Correct Answer: AB
Question #45
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
A. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path
B. Cisco Unified Communications Manager invoked media termination point resources
C. The RTP traffic is arriving beyond the jitter buffer on the receiving end
D. A firewall in the media path is blocking TCP ports 16384-32768
View answer
Correct Answer: D
Question #46
The SIP session refresh timer allows the RTP session to stay active during an active call. The Cisco UCM sends either SIP-INVITE or SIP-UPDATE messages in a regular interval of time throughout the active duration of the call. During a troubleshooting session, the engineer finds that the Cisco UCM is sending SIP-UPDATE as the SIP session refresher, and the engineer would like to use SIP-INVITE as the session refresher.What configuration should be made in the Cisco UCM to achieve this?
A. Change Session Refresh Method on the SIP profile to INVITE
B. Increase Retry INVITE to 20 seconds on the SIP profile
C. Enable Send send-receive SDP in mid-call INVITE on the SIP profile
D. Enable SIP Rel1XX Options on the SIP profile
View answer
Correct Answer: A
Question #47
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-bandDTMF is supported, what is a reason for this malfunction?
A. he negotiated RTP port is outside of the range described by RFC, so inband DTMFs do not work
B. here is SIP Delayed Offer
C. he rtpmap:0 value for the negotiated codec is marking DTMF as inactive
D. o DTMF is negotiated
View answer
Correct Answer: D
Question #48
Which services are needed to successfully implement Cisco Extension Mobility in a standalone Cisco Unified Communications Manager server?
A. Cisco Extended Functions, Cisco Extension Mobility, and Cisco AXL Web Service
B. Cisco CallManager, Cisco TFTP, and Cisco CallManager SNMP Service
C. Cisco CallManager, Cisco TFTP, and Cisco Extension Mobility
D. Cisco TAPS Service, Cisco TFTP, and Cisco Extension Mobility
View answer
Correct Answer: C
Question #49
Configure Call Queuing in Cisco Unified Communications Manager. Where do you set the maximum number of callers in the queue?
A. n the queuing configuration
B. n Cisco Unified CM Service Parameters
C. n the telephony service configuration
D. n Cisco Unified CM Enterprise Parameters
View answer
Correct Answer: A
Question #50
A company is using third-party phones on the internal network at a location behind a Cisco Unified Border Element. These phones seem to work with only two types of codecs, which are high-complexity codecs.Which configuration ensures that calls are completed with the correct and working media each time?
A. Configure a codec preference list on the Cisco Unified Border Element with the voice class codec command, set the codec preference list codecs, and add it to the dial peer
B. Configure a codec priority list on the Cisco Unified Border Element with the voice class codec command, set the codec priority list codecs, and add it to the dial-peer voice
C. Configure a codec preference list on the Cisco Unified Border Element with the VoIP class codec command, set the codec preference list codecs, and add it to the dial peer
D. Configure a codec preference list on the Cisco Unified Border Element with the voice class codec command, set the codec priority list codecs, and add it to the VoIP peer
E. Reveal Answer
View answer
Correct Answer: B
Question #51
A company has an SRST gateway running an IOS XE image. The company plans to enable the IPv6 addressing companywide. To enable the IPv6 in a unified SRST gateway to support SIP phones, what are two supported supplementary features for an IPv6 fallback scenario? (Choose two.)
A. three-way conference
B. secure SIP lines
C. T
D. transcoding
E. SIP trunk
View answer
Correct Answer: AC
Question #52
A voice administrator enabled secure media between multiple Cisco UCM clusters that are connected via a SIP trunk.Which configuration parameter in the SIP trunk must be enabled to allow encrypted media traffic to traverse the SIP trunk?
A. Check the Allow iX Application Media check box in the SIP Profile
B. Check the SRTP Allowed check box in the SIP Trunk configuration
C. Check the RTP Allowed check box in the SIP Trunk configuration
D. Uncheck the RTP Allowed check box in the SIP Trunk configuration
View answer
Correct Answer: B
Question #53
An engineer must configure a Cisco UCM hunt list so that calls to users in a line group are routed to the first idle user and then the next.Which distribution algorithm must be configured to accomplish this task?
A. broadcast
B. top down
C. longest idle time
D. circular
View answer
Correct Answer: C
Question #54
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?
A. The calling transformation on the SIP trunk is applied first, and then the calling transformation on the route patterns is applied
B. Calling transformations cannot be configured on both route patterns and a SIP trunk
C. The calling transformation on the route patterns is ignored because there is already a calling transformation on the SIP trunk
D. The engineer must also apply a calling transformation at the route group details level
E. Reveal Answer
View answer
Correct Answer: D
Question #55
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?
A. midcall-signallng passthru
B. midcall-signaling preserve-codec
C. no update-callerid
D. midcall-signaling passthru media-change
E. Reveal Answer
View answer
Correct Answer: D

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