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CCIE Voice Test

Instructions:

Answer 50 questions in 15 minutes.
If you are not ready to take this test, you can study here.
Match each statement with the correct term.
Don't refresh. All questions and answers are randomly picked and ordered every time you load a test.

This is a study tool. The 3 wrong answers for each question are randomly chosen from answers to other questions. So, you might find at times the answers obvious, but you will see it re-enforces your understanding as you take the test each time.

1. What are the commands to create the L3 routing interface for VLANS (SVI)?
Int vlan 130|ip address 10.10.101.1 255.255.255.0|int vlan 240|ip address 10.10.201.1 255.255.255.0
Device Pool Locations
Find the GUI Files - show flash | i .html|ip http server|ip http path flash:/<PATH TO FILES>||telephony service|web admin system name admin pass cisco|dn-webedit
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.

2. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
Sh ip rsvp reservation||sh sccp connections
SIP Route Pattern over a SIP Trunk.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
#voice service voip| #no supplementary-service h225-notify cid-update

3. What is Mobile Voice Access?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

4. How do you enable AAR?
(config)#isdn switch-type primary-net5|(config)#network-clock-participate wic 0||(config)#Controller e1 0/0/0|(config-controller)#pri-group timeslot 1-3 !!!Get the PRI up!!!|(config-controller)#no shut||(config)#int s0/0/0:15|(config-if)#isdn outgoin
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Master/Slave relationship. CUCM controls it.
Phone number followed by domain name.|i.e. 3006@ipxcme.com

5. What are two caveats to using the ccm-manager config server</ip> command?

Assign a logout-profile to the ephone.
The + character is not supported on a VOIP dial-peer.||So you'll need to add back on the POTS dial-peer before it gets to the PSTN via a voice translation-rule.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

6. How do you configure CUCM redundancy on an H323 gateway?
Dial-peers
#voice class h323 1| #h225 timeout tcp establish 3||#dial-peer voice 3000 voip| #destination-pattern 3...| #voice-class h323 1| #session target ipv4:(SUB IP)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #voice-clas
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Find the GUI Files - show flash | i .html|ip http server|ip http path flash:/<PATH TO FILES>||telephony service|web admin system name admin pass cisco|dn-webedit

7. 'By Nov. 16 - 2010||I am happy confident and thankful that I am now a CCIE Voice Certified engineer! My technical knowledge of the Cisco UC portfolio is truly unsurpassed. I have opened up a wealth of opportunities for myself and my family and
...
#after-hours block pattern
voice translation-profile|and|voice translation-rule
Assign SIP Dial Rules

8. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
1 ... this is not optional!
Voice register dn 1|number 3001|voice register pool 1|number 1 dn 1|id mac 001b.ac02.14de|type 7961|dtmf-relay rtp-nte|description 3214-3005 !!! PHONE BANNER !!!|codec g711u
Bottom up - to prevent glare.
SIP Dial Rules

9. How do you configure a SIP Trunk? (router side)
voice register pool|and|voice register dn
(config)#isdn switch-type primary-ni|(config)#network-clock-participate wic 0||(config)#controller t1 0/0/0|(config-controller)#pri-group timeslots 1-3|(config-controller)#no shut||(config)#voice service voip|(conf-voi-serv)#sip|(conf-serv-sip)#bind
SIP Route Pattern over a SIP Trunk.
Via the voice hunt-group parallel command

10. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
In the CUCM 7.0.1 Release Notes ... search for restore.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
CUCM Service Parameters --> Call Park Reversion Timer

11. Privacy is enabled system-wide in CUCM by default. (T or F)
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
(config)#isdn switch-type primary-net5 (or whatever's provided)|(config)#network-clock-participate wic 0||(config)#controller e1 0/0/0|(config-controller)#pri-group timeslot 1-3|(config-controller)#no shut||(config)#int s0/0/0:15 (this is created aft
SIP Dial Rules
TRUE

12. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Single Number Reach (Mobile Connect)
#test voice translation rule 1 <input to test>
In CUCM - configure CFUR to point to it's E164 number.

13. What are the commands to create vlans on an ESW?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.

Redundant dial plan. One on CUCM and one on the routers via dial-peers.

14. How do you test a Voice Translation Rule?
#test voice translation rule 1 <input to test>
Route Patterns --> Route Lists --> Route Groups --> Gateways/Trunks|RPs point to Route Lists which contain one or more Route Groups which contain one or more Gateways/Trunks.
Precede the # with a > ... so 9011*>#
To enable two-stage dialing.

15. How would you enable security on a GK?
SIP Route Pattern over a SIP Trunk.
Dtmf-relay h245-alpha
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Phone number followed by domain name.|i.e. 3006@ipxcme.com

16. What kind of relationship does an MGCP gateway have with CUCM?
Device Pool Locations
Master/Slave relationship. CUCM controls it.
#after-hours block pattern
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

17. What does Display-IE do?
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Auto Call Pickup Enabled
Sends the Calling Name.

18. What's the best way to perform digit manipulation on a route group?
Via the voice hunt-group parallel command
To enable two-stage dialing.
Configure it on the route group through the route list - then it will be local to the route list.
Debug isdn q931

19. Maximum Wait Time for Desk Pickup?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Matches any length dialed number and truncates it to 4 digits.
CUCM OS Administration Settings --> NTP Servers
Apply it to the ephone.

20. When are digits stripped in a gateway?
voice register pool|and|voice register dn
Mobile Voice Access
#after-hours block pattern
When they are explicitly matched in a destination-pattern in a dial-peer.

21. What are the commands to configure SIP phones in CME?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Voice register global|mode cme|source-address <IP>|max-dn <NUM>|max-pool <NUM>|tftp-path flash:/<PATH>|create profile||NOTE: You have to do a create profile every time you change the voice register pool
Matches any length dialed number and truncates it to 4 digits.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

22. 'How do you inform a SIP phone of NTP information?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
#voice service voip| #no supplementary-service h225-notify cid-update
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Ccm-manager music-on-hold

23. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
The Called/Calling Transformations are superceded.
SERVICE PARAMETER > Device Name of GK0controlled Trunk That Will Use Port 1720 > US_Trunk (or whatever you're using)||> Host Name/IP of GK That Will Use RAS UDP Port 1719 > (IP of GK) ||BE SURE TO RESTART THE TRUNK in CUCM and possibly the CUCM Servi
(config)#isdn switch-type primary-net5|(config)#network-clock-participate wic 0||(config)#Controller e1 0/0/0|(config-controller)#pri-group timeslot 1-3 !!!Get the PRI up!!!|(config-controller)#no shut||(config)#int s0/0/0:15|(config-if)#isdn outgoin
Mobile Voice Access

24. How do you see multicast packets being sent?
Create 2 voip dial-peers - one the Sub and one to the Pub (the Pub with a preference of 1). Then create a voice class h323and set the h225 timeout value so it will hunt between the two.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Debug ephone moh
sh ephone reg|and|debug voip dialpeer

25. On an H323 GW - how do you adjust the timers for redundancy hunting?
Call Simulator. You can use to validate path from router to the PSTN.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
#int lo0|#h323-gateway voip bind srcaddr [IP]||#isdn switch-type primary-ni|#network-clock-particpate wic0||#controller t1 0/0/0|#pri-group timeslots 1-3||#int serial 0/0/0:23|#isdn outgoing display-ie|#isdn b-chan selection order [use opposite of wh

26. How do do configure TEHO?
Create a Route Pattern for the local site and the remote site:||RP-LOC --> PT-REM --> RL-TEHO-LOC||RP-REM --> PT-LOC --> RL-TEHO-REM
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Call forwarding between voip to voip (when CUBE is in play)
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

27. What are the steps to integrate CUPS with CUCM?
On CUCM -- UPTSAL||Create Your Users|Phone - set owner id|Create SIP Trunk|SIP Trunk Security Profile||Application Server|Licensing||||On CUPS||Work Left to right
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Show ccm-manager
Voice register global|mode cme|source-address <IP>|max-dn <NUM>|max-pool <NUM>|tftp-path flash:/<PATH>|create profile||NOTE: You have to do a create profile every time you change the voice register pool

28. How do you configure an MGCP GW? (router side)
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
#test voice translation rule 1 <input to test>
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
(config)#ip domain-name <NAME>|(config)#isdn switch-type primary-ni|(config)#network-clock-participate wic 0||(config)#controller t1 0/0/0|(config-controller)#pri-group timeslots 1-3 - 24 service mgcp|(config-controller)#no shut||(config)#interface s

29. When a CUCM device dials a number - what happens?
8kb/s
It matches a route pattern| Which points to a route list| Which point to a route group| Which contains a GW or Trunk|*Then it needs to match on an incoming VOIP diap-peer (incoming called -number .)
Precede the # with a > ... so 9011*>#
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

30. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

31. Where is it best to manipulate digits for inbound calls?

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32. What would force you to use telephony-service to configure SRST?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'

Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

33. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
voice register pool|and|voice register dn
Create a Route Pattern for the local site and the remote site:||RP-LOC --> PT-REM --> RL-TEHO-LOC||RP-REM --> PT-LOC --> RL-TEHO-REM
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.

34. What are the commands to configure an H323 GW?
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
#int lo0|#h323-gateway voip bind srcaddr [IP]||#isdn switch-type primary-ni|#network-clock-particpate wic0||#controller t1 0/0/0|#pri-group timeslots 1-3||#int serial 0/0/0:23|#isdn outgoing display-ie|#isdn b-chan selection order [use opposite of wh
Assign SIP Dial Rules
Service Parameter --> Block OffNet To OffNet Transfer --> True

35. Describe how you configure SIP URI functionality.
The + character is not supported on a VOIP dial-peer.||So you'll need to add back on the POTS dial-peer before it gets to the PSTN via a voice translation-rule.
To enable two-stage dialing.

At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

36. How do you enable Extension Mobility for a device in CME?
TRUE
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
#isdn switch-type primary-ni|#network-clock-participate wic 0|#controller t1 0/0/0|#pri-group timeslots 1-3 service mgcp||#ccm-manager mgcp ccm-manager redundant-host [IP]|#ccm-manager fallback-mgcp (for srst)|#ccm-manager moh||#mgcp|#mgcp call-agent
Assign a logout-profile to the ephone.

37. What are the 3 mandatory commands within call-manager-fallback?
TRUE
It matches a route pattern| Which points to a route list| Which point to a route group| Which contains a GW or Trunk|*Then it needs to match on an incoming VOIP diap-peer (incoming called -number .)
Max-dn|max-ephone|ip source-address
Mobile Voice Access

38. How do you prevent toll fraud on CUCM?
In CUCM - configure CFUR to point to it's E164 number.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Route Patterns --> Route Lists --> Route Groups --> Gateways/Trunks|RPs point to Route Lists which contain one or more Route Groups which contain one or more Gateways/Trunks.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True

39. What types of digit manipulation can you perform at the route pattern?
Automatically configures the MGCP GW for you.
Create 2 voip dial-peers - one the Sub and one to the Pub (the Pub with a preference of 1). Then create a voice class h323and set the h225 timeout value so it will hunt between the two.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Call forwarding between voip to voip (when CUBE is in play)

40. What does ccm-manager config server [IP]</ip> do?
Use the alias static configuration||alias static (IP of CUCM) gkid (GK ID) gateway voip ras (IP of CUCM) e164 (dn) e164 (dn) e164 (dn)
Automatically configures the MGCP GW for you.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

41. What is Phone NTP Reference used for?
It's best to strip digits at the voice port.
Call forwarding between voip to voip (when CUBE is in play)
To notify SIP Phones of NTP
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

42. How do you configure AAR?
IP Voice Media Streaming App
In the CUCM 7.0.1 Release Notes ... search for restore.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Mobile Voice Access

43. How do you place SCCP and SIP phones into a single huntgroup?
Via the voice hunt-group parallel command
...
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
SERVICE PARAMETER > Device Name of GK0controlled Trunk That Will Use Port 1720 > US_Trunk (or whatever you're using)||> Host Name/IP of GK That Will Use RAS UDP Port 1719 > (IP of GK) ||BE SURE TO RESTART THE TRUNK in CUCM and possibly the CUCM Servi

44. What are the commands to configure a SIP phone in CUCME?
Voice register dn 1| number 4002||voice register pool 1|id mac 0022.0022.0022|type 7961ge|number 1 dn 1|dtmf-relay rtp-nte|no vad
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
...
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

45. How would you set the the timer for Auto Answer?

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46. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?

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47. How do you allow the Calling Name to be sent to the PSTN on a router?
On CUCM it's identical to adding an H323 GW.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Create a Route Pattern for the local site and the remote site:||RP-LOC --> PT-REM --> RL-TEHO-LOC||RP-REM --> PT-LOC --> RL-TEHO-REM
#voice service voip| #no supplementary-service h225-notify cid-update

48. On the RDP What is the Rerouting CSS used for?
Debug isdn q931
Single Number Reach (Mobile Connect)
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
On CUCM -- UPTSAL||Create Your Users|Phone - set owner id|Create SIP Trunk|SIP Trunk Security Profile||Application Server|Licensing||||On CUPS||Work Left to right

49. When using GK redundancy - what must you do to ensure both SUB and PUB use port 1720 for call-signalling.

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50. An MGCP gateway serving as an SRST router requires what?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
(config)#voice register dialplan
24kb/s
#test voice translation rule 1 <input to test>