A phone
has a device CSS that includes the partitions phones and PSTN. This provides
access
for all
internal phones and external calls. The first line on this phone has a CSS that
includes the
partitions
phones, 911, and local. If a call is placed to a long-distance number, will the
call be
completed
and why or why not?
Yes, because the call will use the device CSS
When
creating your own CSV file to be used by Cisco Unified Communications Manager
BAT,
which three
rules do you need to follow?
1. Edit the
Excel macro for each separate Unified Communications Manager BAT transaction.
2. Use a
separate line for each data record.
3. Separate
each data field with a comma and include comma separators for blank fields.
There are
three servers listed in a Cisco Unified Communications Manager group. They are
listed
in this
order: Subscribe_1, Publisher, and Subscriber_2. In which order does an IP
phone attempt
to register
with the Cisco Unified Communications Manager group?
The IP
phone will go through the list from first server to last server listed in the
Cisco Unified
Communications
Manager group list.
Which two
call-routing tools can be applied to a device pool? (Choose two.)
1. Called
and Calling Party Transformation CSS
2. Standard
Local Route Group
What is
required to configure NTP on a Cisco Unified Communications Manager publisher
to
support SIP
phones?
Configure
Phone NTP References in Cisco Unified CM Administration.
Which layer of the Cisco Unified Communications components is responsible for delivering a dial tone?
Call control is the CUCM component responsible for delivering a dial tone.
What is the name of the server in a CUCM cluster that maintains a read/write copy of the entire database?
Publisher is the name of the server in a CUCM cluster that maintains a read/write copy of the entire database.
What protocol is responsible for transporting voice over IP?
Real-Time Transport protocol (RTP) is responsible for transporting voice over IP.
How many call-processing agents can be active in a CUCM cluster?
There are 4 call-processing agents that can be active in a CUCM cluster.
How many servers can be in a CUCM cluster?
There can be 20 servers in a CUCM cluster.
Which CUCM server is the license manager component active on?
The license manager component is active on all servers.
Which CUCM server is the license server component active on?
The license server component is active on the publisher server.
On which server in the CUCM cluster are license files loaded?
Publisher License files are loaded on the publisher server.
Which of the following features is not a user-facing feature (UFF)?
Attendant console (login/logout) is not a user-facing feature (UFF).
What is CUCM Clustering and its types.
The clustering feature of Cisco Call Manager provides a
mechanism for seamlessly distributing call processing across the infrastructure
of a converged IP network. Clustering provides transparent sharing of resources
and features and enables system scalability
1.Single
Site
2.Multisite
with Centralized Call-Processing
3.Multisite
with Distributed Call-Processing
4.Clustering
Over the IP WAN
What is Clustering Over WAN.
3- Difference between MGCP and H323 gateway.
4- Diff between SIP/MGCP/h323
H.323 and SIP are both referred to as “call control” protocols. They allow a device, such as a desk phone, softphone, or videoconferencing system to place a call to another person over IP. 5- Media Resources- how to configure them. Transcoders, MTP.
6- Difference between Location and Region tab, Device pool, Codec.
7- Difference between Translation and Transformation pattern.
8- Feature sets- Hunt Group, Pick up group, shared lines, barge in, Privacy
9- Gatekeeper- Registration msgs, registering gateways via diff technologies, roles of GK/ Gateway- Advantages of MGCP and H323, MGCP backhaul.
10- Extension mobility, QOS
11- Difference between SRST and MGCP Fallback.
12- ISDN, RTP, RTCP etc.
13- What is a partition?
14- What is a CSS?
15- How are partitions and CSSs used in the dial plan?
16- I have all the phone numbers within my organization in the same partition. How can I grant the phone access to call these numbers?
17- My phone has a line CSS and a Device CSS. I have the same exact pattern in a partition in my line CSS and in a partition in my Device CSS. Which pattern takes precedence?
18- So if a call is permitted on the line CSS but blocked on the device CSS is the call routed or blocked?
19- What is Cisco’s Line / Device CSS methodology? If you are not familiar with it no need to answer.
20- The partition which contains all of my organizations phone numbers could either be placed in a CSS on the line or a CSS on the phone (device). Which one should I use and why?
21- What is a Local Route group and how does it simplify an implementation?
22- I want to identify four different classes of service for International, National, Local, Internal Only. Can give me a quick overview as to how I can apply a local Class of Service restriction on a phone?
23- Another engineer tells you that he plans to partition the phones logically according to the physical state (i.e. California) that they are located in. All the phones in the same state will be placed in the same device pool. He then asks you if this is how you would assign phones to a device pool. What do you think?
24- What is the difference between a region and a location?
25- When do you need to use DSPs?
26- Can you have a conference bridge without the use of DSPs?
27- What limitations are there to software based conference bridges?
28- How do I make sure all parties are using G711
29- What is the most obvious first step I should take to troubleshoot media resources?
30- How is a hardware based conference bridge configured and applied to a phone? Start with DSPs in your explanation and end with a phone being able to successfully join a conference bridge using a g.729 codec.
31- My H.323 gateway registration shows as Unknown in call manager. What should I do?
32- I am not sure if my PRI is coming up correctly. Is there a frequently used show command that will allow me to know / see if layer 1, 2 and 3 is currently up and working on my PRI.
33- What output from show ISDN status will allow me to know that my layer 3 connection to the telco has been successful?
34- I am getting TEI_ASSIGNED instead of MULTIPLE_FRAME_ESTABLISHED how might I solve this problem?
35- Which debug command will allow me to see what digits I am sending to or receiving from the telco on a PRI?
36- I need a show command that will allow me to troubleshoot clocking on my PRI. Perhaps I misconfigured my clocking and I might be getting errors on my PRI?
37- I have two different dial peers. One for 911 and another for [2-9]…… another engineer tells me that I need to use the command forward-digits all under my dial peer configurations. Is this command really necessary? Which dial peer will this command affect and what does this command do?
38- How can I tell if my MGCP gateway is registering correctly with Call Manager?
39- My MGCP gateway interfaces will not register with call manager what are some of the things I should check first as common errors people make?
40- Another engineer tells me I need to add ISDN bind-l3 ccm-manager to my gateway configuration. What does the following command do and where is it placed? What happens if I fail to add it?
41- When my MGCP gateway goes into SRST mode I notice that ISDN bind-l3 ccm-manager disappears from my configuration. Why is that?
42- If I have configured my Cisco Voice gateway with MGCP do I need to configure any translation-rules or dial peers on the gateway?
43- I have decided I am not going to use any transformation patterns in CUCM in any of my implementations. My users dial 9 for an outside line followed by 7 digits for local calls. Telco wants 10 digits for local. What then do I need to do in order to assure that TELCO accepts my calls?
44- At what layer of the OSI layer model does SIP signaling take place
45- How does NAT cause problems with SIP signaling? How does a CUBE solve these types of problems?
46- What is the difference between medial flow through and media flow around modes? If I am joining two
different companies into one company which method should I use and why?
47- In Call Manager under Device > Trunk > Trunk Configuration I see a box that says: Media Termination Point Required what happens when I check that box? If I check that box and my calls fail why might this be happening? What is required between trunk endpoints for that box to be left unchecked?
48- I need to implement remote sites taking into consideration the dial plan, site features and options, how features and options match up with how the business uses their phone system. What questions do I need to ask to create a list of items that must be implemented for remote sites? The cluster is already built out so only focus on items that meet business requirements for remote sites. Please limit your response to 3 minutes. Example = pickup groups
49- Difference between h323 / MGCP /SCCP /SIP
50- Difference between CUCM / CME
51- Difference between FXS and FXO
52- Difference between T1/E1 PRI and T1/E1 CAS
53- Difference between 7940 / 7960 Ip Phone
54- What is MOH
55- Which codec is using on LAN / WAN
56- Bandwidth require for Voice Call on WAN
57- How many channels are in physical IP Phone
58- What are the Steps to add a H323 Device
59- What are the Steps to add a MGCP Device
60- What are different process running on call manager for registration.
61- What is new on CCM 4 5 CUCM 6 and 7
62- Is CLID supported by H323 and MGCP
63- Steps to add a Fractional T1 on system.
64- Steps to Configure Extension mobility.
65- Steps to configure IPMA.
66- what are all the Debug commands available for all voice protocols
67- Steps to integrate Unity with Call manager and verify the integration
68- Exchange integration with Unity
69- What is partition and CSS
70- How to integrate UCCX and using which protocol
71- What is cluster
72- What is publisher
73- Difference between SNR and MVA
74- Difference between FXS and FXO
75- Difference between and T1/E1 CAS
76- H323 call flow
77- SIP call flow
78- Early offer and delay offer
79- Types of call processing models in cisco ip telephony
80- Can we have SCCP gateway
81- What are the Steps to add a MGCP Device
82- Difference between call handler and user
83- H323 DTMF relay options
84- Steps to Configure MVA
85- what are all the Debug commands available for all voice protocols
86- Difference between h323 and MGCP
87- Why we need transcoder
88- How to add a user in CM
89- What you know about UCS Servers?
90- What’s new in CUCM 9?
91- Configure VLAN
92- Voice VLAN is trunk or access
93- In ether switch module does it required to put switch mode access?
94- Does VLAN can be made as trunk?
95- IP Phone Bootup Process.
96- SIP firmware updating
97- SIP phone registration what config file download from CUCM?
98- How many channels does E1 have? Framing and Signaling
99- What is VWIC2? What is the use of card type command?
100- In media resources use of “SCCP Local” and “DSP services dspfarm” command? Explain Transcoder, MTP and Conference
Dif. b/w IOS Enhanced and Enhanced in CUCM?
PRI-BACKHAUL in MGCP?
MGCP media negotiations.
What port is used by MGCP gateway to communicate with CA?
Directed call park and call park diff.
Ad-hoc and meet-me conference diff.
VirtualiZation for CUCM? Does CUCM 7.0 supports virtualization?
Cisco Unity Connection Call Flow?
How many voice mail port used in your company?
Inbound call matching dial-peer sequence.
Default dial-peer disadvantages?
COR List
SIP Phone registration in CME
Device pool mandatory fields?
RMCM in UCCX?
Diff. b/w CTI RP and CTI Ports
DB replication types explain them
SIP PRACK message
What is SIP trunk and What is SIP Gateway?
IOS 12.4 and CUE 7.0 in CME router downloaded CUE 8.0 can CUE 7.0 access CUE 8.0 files
Adding custom ringtone in IP phone
CME IP Phone registration process with commands
