Test your basic knowledge |

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 change modes in voice register global?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

Call forwarding between voip to voip (when CUBE is in play)
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST

2. How do you set the inter-digit timeout for SIP phones in CME?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
TRUE
#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
(config)#voice register dialplan

3. How do you disable KPML?
voice register pool|and|voice register dn
Assign SIP Dial Rules
Debug ephone moh
#voice service voip| #no supplementary-service h225-notify cid-update

4. How do you test a Voice Translation Rule?
It's best to strip digits at the voice port.
#interface vlan 130 (voice vlan)| #h323-gateway voip interface| #h323-gateway voip id BR2 ipaddr 10.1.6.1| #h323-gateway voip h323-id UK| #h323-gateway voip bind srcaddr 10.1.130.1
#test voice translation rule 1 <input to test>
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

5. How do you configure an MGCP GW? (router side)
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
To enable two-stage dialing.
(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
Bottom up - to prevent glare.

6. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
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
In the CUCM 7.0.1 Release Notes ... search for restore.
SIP Route Pattern over a SIP Trunk.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

7. What does ccm-manager config server [IP]</ip> do?
(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
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
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.
Automatically configures the MGCP GW for you.

8. How do you configure AAR?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Ip domain-name ipexpert.com|network-clock-participate wic 0|isdn switch-type primary-ni||controller t1 0/0/0|pri-group timeslots 1-3 - 24 service mgcp|no shut||int serial 0/0/0:23|isdn bind-l3 ccm-manager|ccm-manager mgcp|ccm-manager redundant-host 1
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
...

9. What does the Transfer On-hook Enabled Service Parameter do?
Allows you to transfer by only pressing the Transfer button once.
'Use before the ... so XXXX
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
To enable two-stage dialing.

10. How do you place SCCP and SIP phones into a single huntgroup?
Master/Slave relationship. CUCM controls it.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Via the voice hunt-group parallel command
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable

11. On the RDP What is the Rerouting CSS used for?
Put them in a route list.
IP Voice Media Streaming App
Bandwidth
Single Number Reach (Mobile Connect)

12. What are the steps to configure Single Number Reach and Mobile Voice Access?
Precede the # with a > ... so 9011*>#

#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
#voice class h323 1| #h225 timeout tcp establish 3||#dial-peer voice 3000 voip| #destination-pattern 3...| # voice-class h323 1| #session target ipv4:10.1.5.3 (sub)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #pre

13. When do you use translate called? Translate calling?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
To notify SIP Phones of NTP
24kb/s
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

14. In Gatekeeper CAC how do you restrict a specific endpoint?
(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
voice translation-profile|and|voice translation-rule
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

15. Name 4 useful show commands for active calls.
(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
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
#after-hours block pattern
It's best to strip digits at the voice port.

16. Name 2 commands to verify RSVP functionality.
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
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Sh ip rsvp reservation||sh sccp connections

17. '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.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
At the CLI: utils ntp status

18. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
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
Re-enter the mgcp bind control and mgcp bind media commands.
Reroute when there is a WAN Outage.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

19. How do you ensure that G.711 only is used?
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.
Higher
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable

20. If you have an H323 Gateway at a siet where the phones are controlled by CUCM - what do you need from a dial-peer perspective?
In CUCM - configure CFUR to point to it's E164 number.
You need pots dial-peers for incoming calls and you need a voip dial-peer pointing to the CUCM so the H323 GW and connect the call to those phones.
It's best to strip digits at the voice port.
Auto Call Pickup Enabled

21. What does Display-IE do?
Sends the Calling Name.
When configuring TEHO.
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.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

22. How do you see the details of calls coming in and out of the PRI?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Ip domain-name ipexpert.com|network-clock-participate wic 0|isdn switch-type primary-ni||controller t1 0/0/0|pri-group timeslots 1-3 - 24 service mgcp|no shut||int serial 0/0/0:23|isdn bind-l3 ccm-manager|ccm-manager mgcp|ccm-manager redundant-host 1
Debug isdn q931
#after-hours block pattern

23. What's the best way to do digit manipulation on an IOS gateway?

Call forwarding between voip to voip (when CUBE is in play)
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
voice translation-profile|and|voice translation-rule

24. How much bandwidth does a G.729 call including layer 3 require?
24kb/s
CUCM Service Parameters --> Call Park Reversion Timer
(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
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

25. What is CSIM?
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
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
Put them in a route list.
Call Simulator. You can use to validate path from router to the PSTN.

26. SNR is also known as?
#interface vlan 130 (voice vlan)| #h323-gateway voip interface| #h323-gateway voip id BR2 ipaddr 10.1.6.1| #h323-gateway voip h323-id UK| #h323-gateway voip bind srcaddr 10.1.130.1
Mobile Connect
Automatically configures the MGCP GW for you.
Put them in a route list.

27. What is the Enable Enterprise Feature Access Service Parameter used for?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
...
To enable two-stage dialing.
Service Parameters --> Mobile Voice Access Number

28. An MGCP gateway serving as an SRST router requires what?
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.
(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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Redundant dial plan. One on CUCM and one on the routers via dial-peers.

29. How do you block an external call from being transferred back out to the pstn by an internal user?
Assign SIP Dial Rules
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
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
Service Parameter --> Block OffNet To OffNet Transfer --> True

30. On an H323 GW - how do you adjust the timers for redundancy hunting?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Master/Slave relationship. CUCM controls it.
When multiple sites use the same route pattern - do your digit manipulation on a route list.

31. What CM Service needs to be start in Serviceability for MOH to work?
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
At the CLI: utils ntp status
IP Voice Media Streaming App
Show ccm-manager

32. What are the commands to manually configure an MGCP gateway?
The Called/Calling Transformations are superceded.
#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
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie

33. What does KPML do?
#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
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

34. How do do configure TEHO?
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
IP Voice Media Streaming App
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
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable

35. 'If you're not getting a DHCP address from CUCM what then?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
The Called/Calling Transformations are superceded.
Assign a logout-profile to the ephone.

36. How do you configure SRST?
CUCM Service Parameters --> Call Park Reversion Timer

Call-manager-fallback or telephony-service mode SRST
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

37. What are two caveats to using the ccm-manager config server</ip> command?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
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 .)

38. How do you see multicast packets being sent?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
#interface vlan 130 (voice vlan)| #h323-gateway voip interface| #h323-gateway voip id BR2 ipaddr 10.1.6.1| #h323-gateway voip h323-id UK| #h323-gateway voip bind srcaddr 10.1.130.1
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Debug ephone moh

39. How do you configure CUBE?

40. Which takes precedence Device Locations or Device Pool Locations?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Reroute when here is WAN congestion.
Device Pool Locations

41. What are the commands to create the L3 routing interface for VLANS (SVI)?
On CUCM it's identical to adding an H323 GW.
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
Outbound dial-peers
Debug isdn q931

42. How do you enable Extension Mobility for a device in CME?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Assign a logout-profile to the ephone.
sh ephone reg|and|debug voip dialpeer

43. What is Phone NTP Reference used for?
Single Number Reach (Mobile Connect)
To notify SIP Phones of NTP
It's best to strip digits at the voice port.
(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

44. What do you need to do to activate the CME GUI?
...
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
#voice class h323 1| #h225 timeout tcp establish 3||#dial-peer voice 3000 voip| #destination-pattern 3...| # voice-class h323 1| #session target ipv4:10.1.5.3 (sub)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #pre
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.

45. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
(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
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
24kb/s
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

46. What are the base telephony-service commands for CME?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

No POTS dial-peer or no direct-inward-dial on a pots dial-peer

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

48. How do you prevent toll fraud on CUCM?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
The name configuration field in ephone-dn and voice register dn
Higher
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

49. Describe how you configure SIP URI functionality.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
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

50. What is the full E164 format?
#interface vlan 130 (voice vlan)| #h323-gateway voip interface| #h323-gateway voip id BR2 ipaddr 10.1.6.1| #h323-gateway voip h323-id UK| #h323-gateway voip bind srcaddr 10.1.130.1
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Sends the Calling Name.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]