DIAL PLAN

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 Ciscos 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?

ADVANCED DIAL PLAN QUESTIONS

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?

MEDIA RESOURCES
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 conf 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.

VOICE GATEWAY H.323 RELEVANT

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?

MGCP GATEWAY
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?

CUBE QUESTIONS
Easy lead in 32. At what layer of the OSI layer model does SIP signaling take place 33. How does NAT cause problems with SIP signaling? 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?

GENERAL IMPLEMENTATION QUESTIONS

36. 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