- 4.9/5.0
- 240 Questions
- Updated on: 12-Oct-2026
- Implementing Cisco Advanced Call Control and Mobility Services (CLACCM)
- 22405 Prepared
Free Cisco 300-815 Practice Questions 2026 | Implementing Cisco Advanced Call Control and Mobility Services (CLACCM)
Refer to the exhibit.
Within the North American Numbering Plan, gateways located in Ottawa, Canada and
marked as “YOW” are assigned to the Calling Party Transformation CSS NANP_CgPTP,
which contains partition NANP_calling_xforms. What is the calling-party number and the
numbering type if the calling user +1613-555-1234 dials the number?
A.
calling number 613-555-1234 and numbering type “subscriber”
B.
calling number 011-1-613-555-1234 and numbering type “subscriber”
C.
calling number 011613-555-1234 and numbering type “international”
D.
calling number 613-555-1234 and numbering type “national”
calling number 613-555-1234 and numbering type “national”
An administrator is implementing a new dial-plan on Cisco Unified Border Element. The
administrator must ensure that incoming dial-peers are matched based on the IP address
from where the incoming request originates. Which dial-peer configuration should be applied to accomplish this requirement?
A.
dial-peer voice 1 voip
incoming url via
B.
dial-peer voice 1 voip
incoming url request
C.
dial-peer voice 1 voip
incoming called-number
D.
dial-peer voice 1 voip
incoming url to
dial-peer voice 1 voip
incoming url via
A user in location X dials an extension at location Y. The call travels through a QoSenabled
WAN network, but the user experiences choppy or clipped audio. What is the
cause of this issue?
A.
missing Call Admission Control
B.
codec mismatch
C.
ptime mismatch
D.
phone class of service issue
missing Call Admission Control
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.
The RTP traffic is arriving beyond the jitter buffer on the receiving end.
An engineer set up and successfully tested a TEHO solution on the Cisco UCM. PSTN
calls are routed correctly using the IP WAN as close to the final PSTN destination as
possible. However, suddenly, calls start using the backup local gateway instead. What is
causing the issue?
A.
WAN connectivity
B.
LAN connectivity
C.
route pattern
D.
route list and route group
WAN connectivity
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.
Set the correct subnet under Device Mobility Info.
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
interworking between an OOB method and RFC2833 for flow-around calls
Reference: https://www.cisco.com/c/en/us/support/docs/unified-communications/unifiedborder-
element/200412-DTMF-Relay-and-Interworking-on-CUBE.html#anc35
Refer to the exhibit.
An engineer is troubleshooting a call-establishment problem between Cisco Unified Border Element and Cisco UCM. Which command set corrects the issue?
A.
SIP binding in SIP configuration mode
voice service voip sip bind control source-interface GigabitEthernetO/0/0 bind media source-interface GigabitEthernetO/0/0
B.
SIP binding In SIP configuration mode:
voice service volp
sip
bind control source-Interface GlgabltEthernetO/0/1 bind media source-Interface
GlgabltEthernetO/0/1
C.
SIP binding In dial-peer configuration mode:
dial-peer voice 300 voip
voice-class sip bind control source-interface GigabitEthernetO/0/1 voice-class sip bind
media source-interface GigabitEthernetO/0/1
D.
SIP binding in dial-peer configuration mode:
dial-peer voice 100 volp
voice-class sip bind control source-interface GigabitEthernetO/0/0
voice-class sip bind media source-interface GigabitEthernetO/0/0
SIP binding in dial-peer configuration mode:
dial-peer voice 100 volp
voice-class sip bind control source-interface GigabitEthernetO/0/0
voice-class sip bind media source-interface GigabitEthernetO/0/0
When the services key is pressed Cisco Extension Mobility does not show up. What is the cause of the issue?
A.
The URL configured for Cisco Extension Mobility is not correct.
B.
Cisco Extension Mobility Service is not running.
C.
The phone is not subscribed to Cisco Extension Mobility Service.
D.
Cisco Extension Mobility is not enabled in the Phone Configuration Window (Device > Phone)
The phone is not subscribed to Cisco Extension Mobility Service.
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.
Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager
with Phone_C information in the Refer-To section.
B.
Phone_B sends a SIP-REFER message to the Cisco Unified CM 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.
Phone_A sends a SIP-REFER message to the Cisco Unified Communications Manager
with Phone_C information in the Refer-To section.
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.
| Page 6 out of 24 Pages |