Hard 350-801 practice questions

Users at a branch site report choppy, robotic audio only on calls that traverse the WAN during business hours; internal branch-to-branch calls are clear. The WAN router's egress interface shows increasing output drops in the default queue during peak periods, and a packet capture shows RTP packets arriving with high jitter and occasional gaps. DSCP markings on the RTP packets are correctly set to EF at the phones and are preserved end to end. What is the most likely root cause and appropriate fix?

A CUBE running IOS XE receives an inbound INVITE from an ITSP. Two inbound dial-peers are configured: dial-peer 100 has 'incoming uri via 10.1.1.1' matching the provider's source IP, and dial-peer 200 has 'incoming called-number 5551234...'. The INVITE arrives from the ITSP source IP 10.1.1.1 with a called number of 5551234567. Both dial-peers appear to be valid candidates. Which dial-peer will CUBE select to process this inbound call leg, and why?

A CUBE connects an ITSP SIP trunk to CUCM. The provider sends RFC 2833 DTMF, but CUCM's SIP trunk is configured for out-of-band (SIP KPML/unsolicited NOTIFY), and calls otherwise complete with two-way audio. Digits pressed by PSTN callers into the CUCM auto-attendant are not recognized. The engineer wants CUBE to convert the DTMF signaling between the two legs while allowing media to bypass CUBE where possible. Which configuration approach correctly addresses this requirement?

A CUBE has two outbound dial peers to reach the PSTN through separate ITSPs. Dial peer 100 has 'preference 1' and points to ITSP-A; dial peer 200 has 'preference 2' and points to ITSP-B. When ITSP-A returns a SIP 503 Service Unavailable, calls fail immediately instead of rolling over to ITSP-B. The engineer confirms both dial peers match the destination pattern equally. What is the most likely cause and correct fix?

A CUBE connects an internal CUCM cluster to an ITSP SIP trunk. Voice calls work perfectly, but every T.38 fax call fails during the switchover to fax. A debug shows that when CUBE sends the re-INVITE to the ITSP, the SDP media line reads 'm=image 4000 udptl t38', while the ITSP's own documentation states its equipment only accepts the media transport token 'UDPTL' followed by 'TCP/UDPTL' capability and rejects lowercase 'udptl'. You must modify the outbound signaling toward the ITSP without affecting the internal leg. Which configuration approach resolves this?

A CUCM administrator has configured two route patterns in the same partition: 9.[2-9]XXXXXX (a 7-digit local pattern) and 9.[2-9]XXXXXXXXX (a 10-digit pattern). A user dials 9 followed by 5551234, but the phone waits about 15 seconds before the call is finally routed. Users complain about the delay on 7-digit local calls. What is the cause of this behavior, and what best resolves it?

A global enterprise uses globalized call routing with all internal directory numbers and route patterns normalized to +E.164. A user in the Chicago site dials a national number that resolves to +19725551234. The administrator wants the call to egress via the Dallas gateway to take advantage of Tail-End Hop-Off (TEHO), but if the Dallas WAN or gateway is unavailable, the call must fall back to egress out the local Chicago PSTN gateway. Which configuration approach correctly achieves this behavior?

A global enterprise uses globalized call routing in Cisco UCM. All PSTN calls are normalized to +E.164 and routed through a single route pattern that points to a route list containing a Standard Local Route Group. A user in the Singapore site places a call to a local Singapore number. The call correctly matches the global route pattern, but it egresses through the US gateway instead of the Singapore gateway. What is the most likely cause?

A collaboration engineer enables mixed-mode on a CUCM cluster using the Tokenless CTL method (utils ctl set-cluster mixed-mode). After the cluster restarts, secure phones fail to register and report 'Trust List Update Failed' when downloading their configuration. Manual inspection shows phones still hold an older CTL file signed by physical USB security tokens from a previous deployment. What is the root cause of the registration failure?

A user in the Chicago office reports that when they dial the 4-digit extension 4001 for a colleague in the same site, the call is routed to a different device with a conflicting overlapping pattern. Both the local site pattern (partition CHI-Internal) and a corporate pattern (partition CORP-Shared) match the dialed digits. The Chicago phone's line CSS lists partitions in this order: CORP-Shared, CHI-Internal. Which statement best explains the routing behavior and how UCM selects the matched pattern?

An engineer configures a Cisco UCM cluster and needs a specific IP phone line to reach both internal extensions and PSTN numbers. The phone's device-level CSS contains only the 'Internal_PT' partition, while the line-level CSS contains 'Internal_PT' and 'PSTN_PT'. A user reports the phone can call internal extensions but cannot dial any PSTN number. What is the most likely cause and correct behavior of UCM CSS processing here?

A new Cisco IP phone running SIP firmware is deployed at a branch site. The network supports both IPv4 and IPv6, but CUCM is configured for IPv4-only communication on its cluster-wide addressing settings. The phone obtains both an IPv4 and IPv6 address via DHCP and DHCPv6. The phone's Common Device Configuration is set to an IP Addressing Mode of 'IPv6 Only'. The phone fails to register with CUCM, and packet captures show the phone attempting to reach the TFTP and CUCM servers over IPv6 addresses that never respond. What is the MOST likely cause of the registration failure?

A newly deployed Cisco IP Phone 8865 running SIP firmware fails to register with CUCM. Packet captures show the phone downloads its configuration file via TFTP successfully, but it never sends a SIP REGISTER message to the primary CUCM node listed in the config. The phone displays 'Registering' and then reverts to the same node repeatedly. The device is defined in CUCM and associated with the correct device pool. What is the MOST likely cause?

A newly deployed Cisco 8865 phone running SIP firmware boots, obtains an IP address via DHCP, and successfully downloads its configuration file from the CUCM TFTP server. However, the phone displays 'Registration Rejected' and never registers to CUCM. Packet captures show the phone sends a SIP REGISTER, but the CUCM node it targets is a subscriber that is NOT running the Cisco CallManager service. The phone was manually configured with a single CUCM group containing only that subscriber. What is the most likely cause of the registration failure?

A CUCM administrator configures globalized call routing but notices that when an internal user dials a directory URI (user@company.com) from a Jabber client, the call fails while dialing the same user's DN succeeds. SIP route patterns for other external domains work correctly. What is the MOST likely cause of the directory URI call failure?

A remote engineer reports that Jabber successfully logs in through MRA and can search the corporate directory, but calls fail to establish with a secure SIP profile applied. Non-secure phones registering internally work fine. Packet captures on the Expressway-C show the SIP INVITE reaching CUCM, but CUCM logs indicate a failed TLS handshake for the encrypted signaling channel. Which action most directly resolves the secure call failure?

A network engineer is building an LLQ policy on a WAN router to protect UC traffic. Voice RTP is classified into class VOICE and video conferencing RTP into class VIDEO. During periods of congestion, users report that voice remains clear but the video stream occasionally freezes even though there is available bandwidth on the link. Reviewing the policy-map, the engineer sees the VOICE class uses 'priority 1000' and the VIDEO class uses 'bandwidth 2000'. Which statement best explains the behavior and the correct design principle?

During a network audit you review the following QoS policy applied outbound on a WAN interface: a class-map matches voice RTP (DSCP EF), and the policy-map assigns it 'priority 512'. The circuit is congested during peak hours, and users report that after a large batch of new calls connect, some active calls develop choppy audio while others sound fine. Signaling and video classes are unaffected. What is the most likely explanation for this behavior?

A caller on an IP phone reports that when dialing an external number through the CUBE-connected PSTN provider, they hear silence instead of ringback tone until the far end answers. Using 'debug ccsip messages' on CUBE, you observe that the ITSP sends a SIP 183 Session Progress with SDP shortly after the INVITE, but the calling phone never plays local ringback. What is the MOST likely cause of the missing audio during the alerting phase?

Users on a wireless voice VLAN report choppy, garbled audio during calls, while wired phones on the same site are unaffected. RTCP statistics on the affected phones show low packet loss (0.2%) but jitter values fluctuating between 60 ms and 120 ms. The QoS policy marks voice with EF (DSCP 46) end to end, and the WLAN has WMM enabled. Which action is the MOST likely to resolve the choppy audio for the wireless users?

A newly deployed batch of Cisco IP phones fails to register with a CUCM cluster running in mixed mode (secured). Phones that were already registered before enabling mixed mode continue to work normally. Packet captures from a failing phone show the TFTP download of the ITL file completing successfully, but the TLS handshake to CUCM fails with a certificate validation error. The phone's displayed date is January 1, 1970. What is the MOST likely root cause?

A network engineer is troubleshooting intermittent voice quality complaints (choppy audio and jitter) on calls that traverse a SIP trunk between a CUCM cluster and a remote site across a QoS-enabled WAN. Signaling is stable and calls connect normally, but during periods of high data traffic the RTP streams degrade. Packet captures at the WAN edge show that the voice media packets are arriving with a DSCP value of 0 (Best Effort) instead of EF. Signaling packets are correctly marked. Which action is MOST likely to resolve the media degradation?

A company uses Expressway-C and Expressway-E for business-to-business (B2B) video calls. Outbound calls to a partner domain (partner.example.com) fail immediately, while inbound calls from that partner succeed. From the Expressway-C, an administrator can ping the partner's Expressway-E by IP address. A packet capture shows no SIP INVITE ever leaves the local network for outbound attempts. What is the MOST likely cause of the outbound call failure?

You configured a secure SIP trunk between Cisco Unified Communications Manager (CUCM) and a Cisco Expressway-C using TLS. Calls over the trunk fail to establish, and the CUCM SDL trace shows a TLS handshake failure with the reason 'certificate verify failed'. The SIP Trunk Security Profile on CUCM is set to Encrypted with 'X.509 Subject Name' populated. Both nodes' certificates are signed by the same enterprise CA, which is loaded into each server's trust store. What is the MOST likely cause of the failure?

A Webex Calling administrator has configured a multi-location deployment. Users report that when they dial a 4-digit extension that also happens to match the beginning of an enterprise significant number (ESN) pattern used for inter-site abbreviated dialing, calls are being routed to the wrong destination. The administrator wants to understand how Webex Calling decides which match to use when both an internal extension and a dial plan pattern could apply to the dialed digits. In the Webex Calling call routing process, which behavior determines the outcome?