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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. How do you create a trunk on the router side?
voice translation-profile|and|voice translation-rule
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.
8kb/s
Interface f0/0|no ip address|no shut||interface f0/0.10|encap dot1q 10 native --> !!! DATA VLAN !!!|ip address 10.10.100.1 255.255.255.0|interface f0/0.20|encap dot1q 20 --> !!! VOICE VLAN !!!|ip address 10.10.200.1 255.255.255.0

2. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
SIP Dial Rules
(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
Apply voice-class codec
Outbound dial-peers

3. What do you need to do to activate the CME GUI?
...
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)
Call-manager-fallback or telephony-service mode SRST
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

4. How do you see the details of calls coming in and out of the PRI?
Debug isdn q931
Bandwidth
Sh ip rsvp reservation||sh sccp connections
Redundant dial plan. One on CUCM and one on the routers via dial-peers.

5. When do you use translate called? Translate calling?
SIP Dial Rules
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Transfer to external is blocked by default.|To ensure only transfer to internal is allowed - make sure only the following tx pattern is under telephony-service||#telephony service| #transfer-pattern ....

6. How do you ensure that G.711 only is used?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
TRUE
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.

7. Does CUCM support RSVP natively?
SIP Dial Rules
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Top Down means that channel 1 will be the first channel used to place outgoing calls.

8. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?

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

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10. How do do configure TEHO?
8kb/s
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
Higher

11. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
#voice service voip| #no supplementary-service h225-notify cid-update
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
CUCM Service Parameters --> Call Park Reversion Timer
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.

12. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
(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
#after-hours block pattern
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Top Down means that channel 1 will be the first channel used to place outgoing calls.

13. How do you setup AutoRegistration in CUCM?
Bottom up - to prevent glare.
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
(config)#sh cdp neigh detail
Precede the # with a > ... so 9011*>#

14. 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

...
Allows you to transfer by only pressing the Transfer button once.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

15. How do you enable AAR system wide?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Dtmf-relay h245-alpha
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

16. When do you do digit manipulation at the route pattern as opposed to the route list?
Sh ccm-manager
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
#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

17. What are the commands to configure a SIP phone in CUCME?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Top Down means that channel 1 will be the first channel used to place outgoing calls.
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
Dial-peers

18. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
Service Parameter --> Block OffNet To OffNet Transfer --> True
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
voice register pool|and|voice register dn
SERVICE PARAMETER> Automated Alternate Routing Enable > True

19. If you can't use a tech-prefix or default tech-prefix how do you get around the fact that CUCM does not register it's endpoint to GK?
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)
Show ccm-manager
At the CLI: utils ntp status
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

20. What are the basic SCCP Commands fro telephony-service in CUCME?
The name configuration field in ephone-dn and voice register dn
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
...
Ephone-dn 1 <dual / octo>|number 3001|description 3214-3001 !!! PHONE BANNER !!!||ephone 1|mac-address 001b.d4c6.ca99|button 1:1|type 7961|max-calls-per-button 5|busy-trigger-per-button 3

21. Calls coming from CUCM to PSTN need what?
Outbound dial-peers
When they are explicitly matched in a destination-pattern in a dial-peer.
In CUCM - configure CFUR to point to it's E164 number.
Higher

22. How do you support multiple codecs on a dialpeer?
Single Number Reach (Mobile Connect)
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Apply voice-class codec
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

23. On the RDP What is the Rerouting CSS used for?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Single Number Reach (Mobile Connect)
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

24. SNR is also known as?
Service Parameters --> Mobile Voice Access Number
Mobile Connect
(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
(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

25. Describe how you configure SIP URI functionality.

CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.

26. When can you not use a Standard Local Route Group?
Device Pool Locations
When configuring TEHO.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
(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

27. How do you configure CUCM redundancy on an H323 gateway?
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)
(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
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
#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

28. How do you use the # as a string terminator within a SIP Dial Rule?
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
(config)#service dhcp|(config)#no ip dhcp conflict logging|(config)#ip dhcp excluded-address <low IP> <hi IP>|(config)#ip dhcp-pool <name>|(dhcpconfig)#network <network name> </nn or mask>|(dhcpconfig)#domain-name <domain>|(dhcpconfig)#dns-server <IP
Precede the # with a > ... so 9011*>#
#after-hours block pattern

29. What are the base telephony-service commands for CME?
Max-dn|max-ephone|ip source-address
Higher
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

30. Where can you assign the AAR Group?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Ccm-manager music-on-hold
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.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

31. What does Display-IE do?
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
Sends the Calling Name.
Call forwarding between voip to voip (when CUBE is in play)
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

32. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
Reroute when here is WAN congestion.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
SIP Dial Rules
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

33. Is the order of the MRGs in an MRGL significant?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
SIP Dial Rules
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Gatekeeper - zone local GK cisco.com 10.1.6.1 - zone local CUCM cisco.com invia GK outvia GK - zone local BR2 cisco.com invia GK outvia GK

34. What's the difference between AAR and SRST?
Bandwidth
Trunk Port Method (3750 or ESW)||interface fa0/1|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10|switchport voice vlan 20|spanning-tree portfast
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Reroute when here is WAN congestion.

35. Name 4 useful show commands for active calls.
Allows you to transfer by only pressing the Transfer button once.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
In the CUCM 7.0.1 Release Notes ... search for restore.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

36. How do you block an external call from being transferred back out to the pstn by an internal user?
Ccm-manager music-on-hold
Service Parameter --> Block OffNet To OffNet Transfer --> True
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Top Down means that channel 1 will be the first channel used to place outgoing calls.

37. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable

38. How do you see multicast packets being sent?
Debug ephone moh
Outbound dial-peers
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

39. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
1 ... this is not optional!
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
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
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.

40. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
Service Parameter --> Block OffNet To OffNet Transfer --> True
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
SIP Route Pattern over a SIP Trunk.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

41. 'If you want to make changes to any softkeys where do you do it?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
'Service Parameters --> Auto Answer Timer
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
Sh ip rsvp reservation||sh sccp connections

42. Privacy is enabled system-wide in CUCM by default. (T or F)
...
TRUE
Apply it to the ephone.
Allows you to transfer by only pressing the Transfer button once.

43. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
8kb/s
Re-enter the mgcp bind control and mgcp bind media commands.
(config)#sh cdp neigh detail
In the CUCM 7.0.1 Release Notes ... search for restore.

44. What terminology translates to AAR?
Debug isdn q931
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Reroute when here is WAN congestion.
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

45. In Gatekeeper CAC how do you restrict a specific endpoint?
Gatekeeper - zone local GK cisco.com 10.1.6.1 - zone local CUCM cisco.com invia GK outvia GK - zone local BR2 cisco.com invia GK outvia GK
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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

46. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
voice translation-profile|and|voice translation-rule
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
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
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

47. What are the basic ephone-dn and ephone commands?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
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
Call forwarding between voip to voip (when CUBE is in play)
Ephone-dn 1 <dual / octo>|number 3001|description 3214-3001 !!! PHONE BANNER !!!||ephone 1|mac-address 001b.d4c6.ca99|button 1:1|type 7961|max-calls-per-button 5|busy-trigger-per-button 3

48. What does ccm-manager config server [IP]</ip> do?
Automatically configures the MGCP GW for you.
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
Ccm-manager music-on-hold
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

49. How do you configure class of service (CoS) in CUCM? CME?
Service Parameters --> Mobile Voice Access Number
Assign SIP Dial Rules
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

50. How do you enable AAR?
(config)#service dhcp|(config)#no ip dhcp conflict logging|(config)#ip dhcp excluded-address <low IP> <hi IP>|(config)#ip dhcp-pool <name>|(dhcpconfig)#network <network name> </nn or mask>|(dhcpconfig)#domain-name <domain>|(dhcpconfig)#dns-server <IP
When they are explicitly matched in a destination-pattern in a dial-peer.
Max-dn|max-ephone|ip source-address
SERVICE PARAMETER> Automated Alternate Routing Enable > True