Cisco UC - Interview Questions

Some of the questions i have collected from various places on the web, will try to keep this updated....

1. What is a partition?
2. What is a CSS?
3. How are partitions and CSSs used in the dial plan?
4. I have all the phone numbers within my organization in the same partition. How can I grant the phone access to call these numbers?
5. 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?
6. So if a call is permitted on the line CSS but blocked on the device CSS is the call routed or blocked?
7. What is Cisco’s Line / Device CSS methodology? If you are not familiar with it no need to answer.
8. 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?
9. What is a Local Route group and how does it simplify an implementation?
10. 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?
11. 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?
12. What is the difference between a region and a location?
13. When do you need to use DSPs?
14. Can you have a conference bridge without the use of DSPs?
15. What limitations are there to software based conference bridges?
16. How do I make sure all parties are using G711
17. What is the most obvious first step I should take to troubleshoot media resources?
18. 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.
19. My H.323 gateway registration shows as Unknown in call manager. What should I do?
20. 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.
21. What output from show ISDN status will allow me to know that my layer 3 connection to the Telco has been successful?
22. I am getting TEI_ASSIGNED instead of MULTIPLE_FRAME_ESTABLISHED how might I solve this problem?
23. Which debug command will allow me to see what digits I am sending to or receiving from the Telco on a PRI?
24. 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?
25. 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?
26. How can I tell if my MGCP gateway is registering correctly with Call Manager?
27. 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?
28. 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?
29. When my MGCP gateway goes into SRST mode I notice that isdn bind-l3 ccm-manager disappears from my configuration. Why is that?
30. If I have configured my Cisco Voice gateway with MGCP do I need to configure any translation-rules or dial peers on the gateway?
31. 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?
32. At what layer of the OSI layer model does SIP signalling take place
33. How does NAT cause problems with SIP signalling? How does a CUBE solve these types of problems?
34. 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?
35. 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?
36. What are the basic differences between H323 gateways and MGCP gateways?        
37. How does PRI deployment differ between the H323 and MGCP?        
38. What are some of the considerations you need to take when you are developing a dial plan?        
39. What are Key issues you had to look into when you did your migrations?
40. Are you familiar with Media resources?  What is your goal when you set them up?  How do you set them up?
41. Are there any benefits of using Hardware vs. Software based media resources?
42. What is the use of IP voice streaming resources with regards to CM services.  What is it used for?
43. What is the benefit of load balancing phones across the servers in a cluster?        
44. Network Management Tools (RTMT)? Which ones have you used? Explain
45. What do you use to communicate between CME and CUE? What is the protocol connection is used to communicate between the CME and CUE?
46. Cisco Services Framework?  Mock integration Experience?
47. CUPSC integration? Any experience?