300-815 Mock Test Free – 50 Realistic Questions to Prepare with Confidence.
Getting ready for your 300-815 certification exam? Start your preparation the smart way with our 300-815 Mock Test Free – a carefully crafted set of 50 realistic, exam-style questions to help you practice effectively and boost your confidence.
Using a mock test free for 300-815 exam is one of the best ways to:
- Familiarize yourself with the actual exam format and question style
- Identify areas where you need more review
- Strengthen your time management and test-taking strategy
Below, you will find 50 free questions from our 300-815 Mock Test Free resource. These questions are structured to reflect the real exam’s difficulty and content areas, helping you assess your readiness accurately.
When locations-based Call Admission Control denies the call, which two masks can AAR apply when routing the call through the PSTN? (Choose two.)
A. AAR destination mask
B. called party transform mask
C. external phone number mask
D. +E.164 alternate number mask
E. enterrise alternate number mask
What is a component of Cisco Unified Mobility?
A. Unified IVR
B. Mobile Connect
C. Smart Client Support
D. Single Number Connect
Customers that call into a company’s IVR report that when they try to select an option, none of the prompts work. The administrator determines that the calls are coming in across an H.323 gateway. While analyzing the dial peer that points toward Cisco UCM, the administrator notices that no DTMF method is configured. Which command resolves this issue?
A. dial-peer voice 2 voipdtmf-relay sip-kpml
B. dial-peer voice 2 voipdtmf-relay h245-alphanumeric
C. dial-peer voice 2 voipdtmf-relay sip-notify
D. dial-peer voice 2 potsdtmf-relay h245-alphanumeric
An administrator is configuring a SIP trunk to an ITSP. The SIP connection will traverse from a Cisco UCM to the ISTP through a Cisco Unified Border Element. The ITSP has indicated that they require an in-band method for DTMF. Which command on the outbound dial-peer to the ITSP will meet this requirement?
A. router (config-dial-peer) dtmf-relay sip-notify
B. router (config-dial-peer) dtmf-relay sip-kpml
C. router (config-dial-peer) dtmf-relay h245-alphanumeric
D. router (config-dial-peer) dtmf-relay rtp-nte
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
An administrator is configuring Meet-me conferencing in a Cisco UCM deployment and has created the Meet-me number and ensured that it is in a partition accessible by all devices. Which two additional steps must the administrator perform? (Choose two.)
A. Ensure that conferencing-initiating devices are using a media resource group list that contains at least one Cisco UCM conference bridge.
B. Disable Early Media on the SIP profile of all devices that will use Meet-me conferencing.
C. Enable Meet-me conferencing in enterprise parameters.
D. Ensure that all devices have G.729 enabled.
E. Update the softkey template on all phones to ensure that they contain the Meet-me softkey.
Refer to the exhibit. Outbound calls to the service provider cause intermittent errors due to a codec mismatch. The internal network sends early offer SDP that contains only G.711 A-law. The service provider reports that some destinations support only G.711 A-law while others support only iLBC. The service provider also allows only 20 active calls at a time. Which configuration allows successful media negotiation for all calls using outbound dial peers 5002 and 5003?
A. dial-peer voice 5002 voipcodec g711alaw ilbc!dial-peer voice 5003 voipcodec g711alaw ilbc
B. dial-peer voice 5002 voipvoice-class codec 101 offer-all!dial-peer voice 5003 voipvoice-class codec 101 offer-all
C. dial-peer voice 5002 voipvoice-class codec 101!dial-peer voice 5003 voipvoice-class codec 101
D. dial-peer voice 5002 voipcodec g711alaw!dial-peer voice 5003 voipcodec ilbc
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.)
A. Enanle PRACK.
B. Enable Early Offer on the Cisco Unified Border Element.
C. Enable the supplementary-service media-renegotiate command.
D. Enable Media Flow Around
E. Enable Mid-Call Signaling Consumption.
Refer to the exhibit. A call made through the Cisco Unified Border Element to pilot 2000 is failing. What is causing the call to fail?
A. The Cisco Unified Border Element is not receiving a response to its OPTION keepalives.
B. The destination pattern is incorrect for the dialed number.
C. VAD was not disabled on the outgoing dial peer.
D. No codecs are configured on the dial peers.
Which two statements are correct with respect to the Client Matter Code setting in the route pattern configuration? (Choose two.)
A. 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.
B. If you check the Allow Overlap Sending check box, the Require Client Matter Code check box becomes disabled.
C. If you check the Allow Overlap Sending check box, you can also check the Require Client Matter Code check box.
D. 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.
E. The Client Matter Code has the option to configure Authorization Level such as in the Forced Authorization Code.
Users report strange behaviors when they call external numbers from the company phone system. The company is using Cisco 8865 phones with Cisco UCM connected to a gateway with two TI circuits. There is no indication of the possible problem, but the connection seems to be established. Which command meets the requirement to capture all the data needed to debug the problem?
A. debug mgcp media tracelevel critical
B. debug mgcp media tracelevel critical all
C. debug mgcp media tracelevel critical verbose
D. debug mgcp media tracelevel moderate
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?
A. rule 1 /^/+([^1].*)/ /0111/
B. rule 1/^+1([2-9]..[2-9]”¦”¦$)/ /1/
C. rule 1 /^([2-9]..[2-9]”¦”¦$)/ /1/
D. rule 1 /^+1([2-9]..[2-9]”¦”¦$)/ //
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?
A. interworking between an OOB method and RFC2833 for flow-around calls
B. interworking between h245-signal and rtp-nte
C. interworking between an OOB method and RFC2833 for flow-through calls
D. interworking between h245-alpha numeric and sip-kpml
DRAG DROP - Drag and drop the steps from the left into the order to provision mobility users through LDAP on the right. Not all options are used.
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. DTMF is supported only in Early Offer.
C. The rtpmap:0 value for the negotiated codec is marking DTMF as inactive.
D. No DTMF is negotiated.
An administrator is configuring a new deployment using Cisco Unified CME. The SCCP phones register without any issues, but SIP phones are not registering. Assume that all other configuration is valid. Which code allows SIP phones to register to Cisco UCME?
A. voice service voipallow-connections sip to h323
B. voice service voipsipbind media source-interface Vlan100
C. voice service voipsipbind control source-interface Vlan100
D. voice service voipsipregistrar server expires max 600 min 60
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.
Where on Cisco Unified Communications Manager do you configure the standard local route group for a group of devices?
A. System > Location Info
B. Call Routing > Route/Hunt > Local Route Group Names
C. System > Device Pool
D. Call Routing > Emergency Location > Emergency Location (ELIN) Groups
You see the voice register pool 1 command in your Cisco Unified Communications Manager Express configuration. Which configuration is occurring in this section?
A. configuration for a single SIP phone
B. configuration items common for all SIP phones
C. configuration for a pool of SIP phones (similar to device pool on Cisco Unified Communications Manager)
D. configuration for SIP registrar service
Which description of RTP timestamps or sequence numbers is true?
A. The sequence number is used to detect losses.
B. Timestamps increase by the time “carrying” by a packet.
C. Sequence numbers increase by four for each RTP packet transmitted.
D. The timestamp is used to place the incoming audio and video packets in the correct timing order (playout delay compensation).
55697959.007 |12:20:50.913 |AppInfo |RouteListCdrc::createPartyTransformedCcSetupReqMsg - before DAapplyCdpnXform() preXformCdpn=11112222 preTag=SUBSCRIBER prePos=11112222 crCdpnMask=33334444 crPrefixDigit= crDDI=2 55697959.008 |12:20:50.913 |AppInfo |RouteListCdrc::createPartyTransformedCcSetupReqMsg - after DAapplyCdpnXform() xformCdpn=33334444 xformTag=SUBSCRIBER xformPos=11112222 55697959.009 |12:20:50.913 |AppInfo |RouteListCdrc::transformed cdpn (without unconsumpt digits) = 33334444, unconsumed digit= Refer to the exhibit. Which INVITE is sent to 10.10.100.123 as a result of this log?
A. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp – wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41[95992364,NET]INVITE sip:33334444@10.10.100.123:5060 SIP/2.0Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3aeFrom: “1000” ;tag=32412716~41f7To: Date: Thu, 01 Apr 2021 17:20:50 GMTCall-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50Supported: timer,resource-priority,replacesMin-SE: 1800 -User-Agent: Cisco-CUCM12.0 –
B. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp – wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41[95992364,NET]INVITE sip:33334444@10.10.100.123:5060 SIP/2.0Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3aeFrom: “11112222” ;tag=32412716~41f7To: Date: Thu, 01 Apr 2021 17:20:50 GMTCall-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50Supported: timer,resource-priority,replacesMin-SE: 1800 -User-Agent: Cisco-CUCM12.0 –
C. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp – wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41[95992364,NET]INVITE sip:11112222@10.10.100.123:5060 SIP/2.0Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3aeFrom: “1000” ;tag=32412716~41f7To: Date: Thu, 01 Apr 2021 17:20:50 GMTCall-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50Supported: timer,resource-priority,replacesMin-SE: 1800 -User-Agent: Cisco-CUCM12.0 –
D. 55698034.001 |12:20:50.922 |AppInfo |SIPTcp – wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.100.123 on port 5060 index 41[95992364,NET]INVITE sip:11112222@10.10.100.123:5060 SIP/2.0Via: SIP/2.0/TCP 10.122.200.50:5060;branch=z9hG4bK268d6e4e48f3aeFrom: “11112222” ;tag=32412716~41f7To: Date: Thu, 01 Apr 2021 17:20:50 GMTCall-ID: 99878a80-66100f2-265e57-67071d0a@10.122.200.50Supported: timer,resource-priority,replacesMin-SE: 1800 -User-Agent: Cisco-CUCM12.0
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.
The sales department must answer phones when other sales members are not at their desks. The administrator knows that configuring Call Pickup allows the sales users to answer all the calls in the department by pressing only the softkey. Which call pickup configuration meets this requirement?
A. Standard Call Pickup
B. Group Call Pickup
C. Other Group Call Pickup
D. Directed Call Pickup
A customer routes PSTN calls to ITSP through a SIP trunk on Cisco UCM that forwards and receives calls to and from ITSP. ITSP is set to send an E.164 number when the customer’s extension is four digits. Which action should be taken to route the incoming calls to four-digit extensions?
A. Configure a voice translation profile to map the E.164 number to four digits and assign it to the incoming dial-peer on Cisco Unified Border Element.
B. Set the Significant Digits to 8 on the SIP trunk.
C. Set the Significant Digits to 4 on the SIP trunk.
D. Configure a voice translation rule to map the E.164 number to four digits and assign it to the incoming dial-peer on Cisco Unified Border Element.
An engineer must route all SIP calls in the form of @example.com to the SIP trunk gateway corporate local. Which two SIP route patterns can be used to accomplish this task? (Choose two.)
A. example.com@gateway.corporate.local
B. *@example.com
C. gateway.corporate.local
D. example.com
E. *.*
Refer to the exhibit. A standard local route group is configured for long-distance calls. Calls from building A succeed, but calls from building B fail. On the system, each building has its own device pool. The DNA tool is used to test the configuration. How is this issue resolved?
A. Change the partition of the route pattern.
B. Add a sip trunk inside route group Standard Local Route Group.
C. Modify the route pattern to add a prefix of 91.
D. Add a local route group on the device pool configuration.
Where is the dtmf-relay command configured on Cisco Unified Border Element?
A. in the voice-class VoIP configuration
B. in the VoIP dial peer
C. in global SIP configuration
D. in the VoIP or POTS dial peers
An engineer is troubleshooting an intersite call between two endpoints where the call fails and the message “Not Enough Bandwidth” is displayed. G.729 codec is in use on both sites. First, calls are being properly routed, and the issue happens after the third call is established and the bandwidth utilization between the two sites is under 50%. Which configuration in Cisco UCM must be adjusted to resolve the issue?
A. transcoder
B. location
C. route pattern
D. translation pattern
Which section under the Real-Time Monitoring Tool allows for reviewing the call flow and signaling for a SIP call in real time?
A. Analysis Manager > Inventory > Trace File Repositories
B. System > Tools > Trace and Log Central
C. Voice/Video > Session Trace Log View > Real Time Data
D. Voice/Video > Session Trace Log View > Open From Local Disk
The Cisco Unified Communications Manager Dialed Number Analyzer allows analysis of calls from which two devices? (Choose two.)
A. translation patterns
B. device pools
C. CTI ports
D. CTI route points
E. IP phones
DRAG DROP - Drag and drop the commands from the bottom to the blanks in the code to implement a translation rule to allow only 11 digits to be received over a SIP trunk to a SIP provider. The Cisco UCM is currently sending calls to the Cisco Unified Border Element in E.164 format. Not all options are used.
An administrator deployed a third-party H.323 gateway in a voice environment, but users report call failures when using features like call hold or call transfer. What are two reasons that these features fail? (Choose two.)
A. The CSS of the transfer initiating line does not contain the partition of the supplementary feature extension (DirectTransfer or MoH Number).
B. The MTP that is configured for use within the H.323 gateway configuration is configured as a trusted source, but the third-party gateway does not trust the signing root CA certificate of the MTP certificate.
C. The MTP does not support the negotiated codec, and media renegotiating during the call is not supported.
D. The Media Resource Group List of the H.323 gateway contains only transcoders and conference bridges but no MTP.
E. The third-party gateway does not support supplementary features, so Media Termination Point (MTP) must be inserted.
An engineer must implement call restriction to toll-free numbers using a class of restriction in a branch Cisco UCME. In which two places is the corlist incoming or cor incoming command configured? (Choose two.)
A. “voice register pool ” configuration mode
B. “ephone-dn ” configuration mode
C. “dial-peer cor custom ” configuration mode
D. “voice register global ” configuration mode
E. “telephony-service ” configuration mode
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?
A. Change the destination-pattern on the outgoing dial peer to match “444333222”.
B. Set up translation-profile on the incoming dial peer to match incoming traffic.
C. Create specific matching for “222333444” on the incoming dial peer.
D. Fix the voice translation-rule to match specifically number “222333444” and change it to “444333222”.
An administrator has configured two route patterns, 9.911 and 9.[2-9]XXXXXX. When a user dials 9911. Cisco UCM waits for the T302 timer before routing the call. How will the administrator force interdigit timeout and route the call as soon as the user has finished dialing 9911, without waiting for the T302 timer to expire?
A. decrease the T302 timer in Service Parameters from the default value
B. enable Urgent Priority on the 9.[2-9]XXXXXX pattern
C. enable Urgent Priority on the 9.911 pattern
D. enable Device Override on both route patterns
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?
A. Configure a Route Group and a Route List to send calls with prefix 9713 to the secondary gateway.
B. 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.
C. Configure two CSSs, then add a translation pattern with the secondary CSS to send calls with 9713 to the secondary gateway.
D. Create a Standard Route Group to dynamically route calls with prefix 9713 to the secondary gateway.
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.)
A. Assign the uploaded audio file to “Network Hold MOH Source & Announcements” under Hunt Pilot’s Queuing section.
B. Upload the audio file in “TFTP File Management” via OS Administration GUI.
C. Assign the uploaded audio file to the hunting Line Group member’s “User Hold MOH Audio Source”.
D. Assign the uploaded audio file to the hunting Line Group member’s “Network Hold MOH Audio Source”.
E. Upload the audio file in “MOH Audio File Management” via CM Administration GUI.
When route patterns are defined as precisely as possible on a Cisco UCM, the control and reliability of the calling plan is increased. A route pattern must be added to support calls to this number range: 9135200 to 9135205. The calls are on-net and no translation patterns are configured. Which configuration meets these requirements?
A. 913520[012345]
B. 913520X
C. 913520[1–5]
D. 913520!
A new Cisco UCM deployment is configured, and now the customer requires a set of 10 DNSs to park the calls starting with 8. The users must use the softkey to park the call. Which configuration must be performed to achieve this requirement?
A.
B.
C.
D.
Refer to the exhibit. ILS has been configured between two hubs this configuration. The hubs appear to register successfully, but ILS is not functioning as expected. Which configuration step is missing?
A. Use TLS Certificates must be selected.
B. The Cluster IDs have not been set to unique values.
C. A password has never been set for ILS.
D. Trust certificates for ILS have not been installed on the clusters.
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?
A. Change the “Maximum Wait Time for Desk Pickup” setting from 4 to 6
B. Change the “Maximum Wait Time for Mobility Pickup” setting from 4 to 6.
C. Change the “Maximum Wait Time for Mobility Pickup” setting from 4000 to 6000.
D. Change the “Maximum Wait Time for Desk Pickup” setting from 4000 to 6000.
When configuring hunt groups, where do you add the individual directory numbers that will be part of the group?
A. route group
B. line group
C. hunt list
D. hunt pilot
Which set of commands binds SIP media and signaling to interface GigabitEthernet0/0 when dial peer 1 is chosen for call routing?
A. dial-peer voice 1 voipvoice-class source interface GigabitEthernet0/0
B. voice service voipbind sip source-interface GigabitEthernet0/0
C. voice service voipsipbind all source-interface GigabitEthernet0/0
D. dial-peer voice 1 voipvoice-class bind control source-interface GigabitEthernet0/0voice-class sip bind media source-interface GigabitEthernet0/0
If all patterns below are configured in Cisco Unified Communications Manager which would be used when dialing the pattern "123"?
A. 12!
B. 12X (urgent priority set)
C. 1XX (urgent Priority Set)
D. 12[2-5]
Calls to a particular extension are not routing to voicemail. The user reaches the voicemail system from the handset by pressing the Messages button. Which configuration parameter causes this problem?
A. The voicemail pilot number for call forwarding is missing from the ephone-dn.
B. The voicemail pilot number is missing from the telephony service configuration on Cisco UCME.
C. The voicemail pilot number is missing from the call handling on Cisco Unity Express.
D. The voicemail pilot number for call forwarding is missing from the ephone.
Cisco UCM has 100,000 entries in the database learned through the ILS Service. Parameter ILS Max Number of Learned Objects in Database value is set to 100,000. What will happen to learned data when the service parameter value is reduced to 50,000?
A. Cisco UCM does not write additional ILS learned objects to the database and will delete the last 50,000 entries learned to keep it to the service parameter value.
B. Cisco UCM does not write additional ILS learned objects to the database and keeps the existing database entries.
C. Cisco UCM will overwrite an entry for newly learned data and keep the parameter value at 100,000.
D. Cisco UCM does not write additional ILS learned objects to the database and will delete the first 50,000 entries learned to keep it to the service parameter value.
In Cisco Unified Communications Manager, which tool do you use to check SIP traces?
A. MTP
B. CCSIP
C. RTMT
D. OS Administration Page
Refer to the exhibit. For long-distance calls, users must prefix their dialed number with “91”. The translation pattern was created to strip the 91 as the PSTN expects a 10-digit number. The PSTN also requires the calling number to be set to 9195551234. However, the service provider has said calls with a different calling number are being received. How is this issue resolved?
A. Change the partition of the translation pattern from none to pstn_pt.
B. Disable Use Calling Party’s External Phone Number Mask on the route pattern.
C. Enable Force Authorization Code on the route pattern.
D. Enable Use Calling Party’s External Phone Number Mask on the translation pattern.
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”.
An administrator troubleshoots call failure in a new deployment and finds that the SIP INVITE messages sent to the service provider contain a diversion header with the user's 4-cigit directory number. These 4-digit directory numbers range from 1000 to 9899. The service provider is rejecting the calls because it requires that the diversion header contain 10 digits. Which command on the Cisco Unified Border Element resolves this issue for all users?
A. voice class sip-profiles 105request INVITE sip-header Diversion modify ““
B. voice class sip-profiles 105request INVITE sip-header Diversion modify ““
C. voice class sip-profiles 105request INVITE sip-header Diversion add ““
D. voice class sip-profiles 105request INVITE sip-header Diversion modify ““
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