A customer is using a SIP trunk to route calls to ITSP to decrease the possibility of downtime, the customer invested in a failover device How does the customer ensure reachability to ITSP, so that if one device on ITSP fails, the calls will be routed to another device?
A solution for a large company is being reconfigured to optimize for cost saving. The company has an extensive global QoS-enabled network with enough bandwidth to create a converged network. Local calls are relatively inexpensive in countries the company have operations, but long distance and international calls are expensive.
Which type of configuration supported by Cisco UCM would help optimize cost control for this company?
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 is own device pool. The DNA tool is used to test the configuration. How is this issue resolved?
Some users report poor quality when they travel to another continent and make calls. This issue applies only to one continent and not to others, where typically the dialing is fast and quality is clear. Users experience the same result at home when they call the same numbers in that specific continent. It seems like some users do not exist in the correct PSTN gateway when making calls to a specific country. The company is using TEHO to save on the cost of international or long-distance calling.
They are also using a globalized dial plan. What is the cause of the issue?
CollabCorp is a global company with two clusters, emea.collab corp and apac.collab.corp. URI dialing is implemented and working in each cluster. The company configured routing between clusters to make inter-cluster calls via URI. but this is not working as expected. Which two configuration elements should be checked to resolve this issue? (Choose two.)
For s SIP to SIP call flow, when does Cisco Unified Border Element require transcoding resources for DTMF?
The same digits sent in OOB, as well as in-band on the Cisco Unified Border Element, are being interpreted as duplicate digits by the receiving end. Which configuration must an engineer implement to resolve this issue?
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 company is using Cisco Jabber on-premises to make B2B calls on video. The calls are using Cisco Expressway-C and Expressway-E and have been configured in Cisco UCM to be able to call any URI on the internet. The Jabber client also has voice enabled and must be able to call local, regional, and international numbers. Where must Cisco UCM be configured to meet this requirement for URI dialing?
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?
TESTED 30 Oct 2025
