Free Cisco 210-060 Exam Practice Questions & Explanations
Prepared & Reviewed by the ValidExamDumps Editorial Team
At ValidExamDumps, we consistently monitor updates to the Cisco 210-060 exam questions by Cisco. Whenever our team identifies changes in the exam questions, objectives, focus areas or requirements, We immediately update our exam questions for both PDF and online practice exams. This commitment ensures our customers always have access to the most current and accurate questions. By preparing with these up to date and 100% exam domain coverage questions, our customers can successfully pass the Cisco Implementing Cisco Collaboration Devices exam on their first attempt without needing additional materials or study guides.
Other certification materials providers often include outdated or removed questions by Cisco in their 210-060 exam. These outdated questions lead to customers failing their Cisco Implementing Cisco Collaboration Devices exam. In contrast, we ensure our questions bank includes only precise and up-to-date questions. Our main priority is your success in the Cisco 210-060 exam, not profiting from selling obsolete exam questions in PDF or Online Practice Test.
An administrator wants to deploy a Cisco Jabber softphone for a user who already has a desk phone. The administrator creates a CSF device, associates it with the end user, and adds the same directory number as the desk phone with a shared line configuration.
Statement: With this configuration, the user can answer an incoming call to that directory number on either the desk phone or Cisco Jabber, and the call can be resumed on the other device.
True. When the same directory number is configured on multiple devices in the same partition, CUCM treats it as a shared line. Both the desk phone and the CSF (Jabber) device ring, either can answer, and because it is a shared line the call appears as remote-in-use on the other device and can be resumed there. This is a standard way to deliver softphone plus desk phone functionality for a single user.
A new employee has been created in Cisco Unified Communications Manager and a phone has been added, but when the user logs into Cisco Unified Communications Self Care Portal they cannot see or modify the phone's speed dials.
What is the most likely cause?
For an end user to manage a phone from the Self Care Portal, the device must appear in the Controlled Devices list of the end user configuration (Device Associations). Without that association, the portal shows no devices to configure.
B is wrong because a missing DN would prevent calling, not portal visibility of the device. C actually helps rather than hinders and by itself does not create the association in all cases; it is not a cause of failure. D is wrong because LDAP-synchronized users can be associated with devices and use the Self Care Portal normally.
An administrator is deploying voice messaging for 400 users and needs a solution that also provides integrated voice recognition, a web inbox, and single inbox integration with Microsoft Exchange.
Which Cisco Unified Communications component should be deployed?
Cisco Unity Connection is the voice messaging platform that provides voicemail, integrated speech recognition (voice commands), the Web Inbox, and Single Inbox synchronization with Microsoft Exchange.
The IM and Presence Service provides instant messaging and presence, not voicemail. Cisco Unified Attendant Console provides operator/receptionist call handling. Cisco Emergency Responder tracks phone location for E911 calls. None of those provide voice messaging.
Users at a branch office report that during peak hours calls sound choppy and words are clipped, although calls still connect normally. The WAN link is heavily used for data backups during the same period.
Which two actions best address the quality issue? (Choose two.)
Choppy audio and clipping are classic symptoms of packet loss and jitter caused by congestion. Applying QoS with low-latency queuing gives voice RTP strict priority so it is not delayed or dropped behind bulk data such as backups. Using regions to force a lower bandwidth codec (G.729) across the WAN reduces the bandwidth each call consumes, easing congestion.
B is irrelevant; signaling protocol choice does not fix media congestion. D is incorrect because jitter buffers exist on the endpoints, not on the CUCM server. E has no effect on voice quality and would remove useful discovery and power negotiation information.
A technician is troubleshooting a call between two Cisco IP phones registered to the same Cisco Unified Communications Manager cluster. The phones are in different sites connected by a WAN link. During the call, the technician captures traffic on the WAN and sees RTP streams flowing directly between the two phone IP addresses, but no signaling traffic.
Which statement best explains this behavior?
In a Cisco Unified Communications deployment, call control signaling (SCCP or SIP) is exchanged between each endpoint and Cisco Unified Communications Manager, while the media stream (RTP) is established directly between the two endpoints after CUCM passes the IP address and port information. That is why the WAN capture shows RTP but not signaling, because the signaling goes to the CUCM subnet, not across the phone-to-phone path.
B is incorrect because SRST only activates when the phones lose connectivity to CUCM. C is wrong because CUCM does not relay media in a standard call. D is wrong because an MTP would terminate and re-originate the media at the MTP address, not produce phone-to-phone RTP.