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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. When do you use translate called? Translate calling?
Mobile Connect
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Sends the Calling Name.
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

2. Does CUCM support RSVP natively?
In CUCM - configure CFUR to point to it's E164 number.
TRUE
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.

3. How do you configure 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
Assign SIP Dial Rules
Outbound dial-peers
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.

4. 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
SIP Dial Rules
To enable two-stage dialing.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
...

5. What are the 3 mandatory commands within call-manager-fallback?
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
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Max-dn|max-ephone|ip source-address

6. Where do you use VIA zone?
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
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

7. SNR is also known as?
Mobile Connect
Debug ephone moh
Bottom up - to prevent glare.
Allows you to transfer by only pressing the Transfer button once.

8. How do you verify where MOH is being served up from?

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9. What are the commands to configure a MGCP Gateway? (router)
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
...
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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

10. What does the Transfer On-hook Enabled Service Parameter do?
#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
Allows you to transfer by only pressing the Transfer button once.
On CUCM it's identical to adding an H323 GW.
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

11. What should you configure before entering auto qos?
#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
Sh ccm-manager
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Bandwidth

12. How do you enable Extension Mobility for a device in CME?

AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Assign a logout-profile to the ephone.
Re-enter the mgcp bind control and mgcp bind media commands.

13. What is CSIM?
Call Simulator. You can use to validate path from router to the PSTN.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
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
#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

14. How do you configure phone ports on an ESW?
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
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
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

15. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
Higher
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
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.
Device Pool Locations

16. How do you see the details of calls coming in and out of the PRI?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Debug isdn q931
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
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

17. What CUCM services should you activate?
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 translation-profile|and|voice translation-rule
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
24kb/s

18. 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
Reroute when here is WAN congestion.
Assign SIP Dial Rules
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

19. What are the commands to configure a T1/E1 PRI?
CUCM OS Administration Settings --> NTP Servers
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)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
(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

20. How do you get around relying on DNS for your CUCMs?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Sh ip rsvp reservation||sh sccp connections

21. How do you support multiple codecs on a dialpeer?
8kb/s
Bandwidth
Apply voice-class codec
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.

22. How much bandwidth does a G.729 call including layer 3 require?
24kb/s
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

23. When a CUCM device dials a number - what happens?
When configuring TEHO.
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 .)
SIP Dial Rules
Precede the # with a > ... so 9011*>#

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

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25. How do you create a trunk on the switch side?
Service Parameters --> Mobile Voice Access Number
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
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.

26. How do you setup AutoRegistration in CUCM?
Assign SIP Dial Rules
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
SIP Dial Rules
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

27. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
sh ephone reg|and|debug voip dialpeer

28. How do you prevent toll fraud on CME?
Sh ip rsvp reservation||sh sccp connections
When they are explicitly matched in a destination-pattern in a dial-peer.
(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
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 ....

29. What's the best way to do digit manipulation on an IOS 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)
Automatically configures the MGCP GW for you.
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.

30. What does Display-IE 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
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Sends the Calling Name.

31. How do you configure CUCM redundancy on an H323 gateway?
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
#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
Call-manager-fallback or telephony-service mode SRST

32. What is the Enable Enterprise Feature Access Service Parameter used for?
To enable two-stage dialing.

At the CLI: utils ntp status
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

33. How would you enable security on a GK?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

34. What are the basic SCCP Commands fro telephony-service in CUCME?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
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
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

35. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
#after-hours block pattern
Bottom up - to prevent glare.

36. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
When configuring TEHO.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
SIP Route Pattern over a SIP Trunk.
Master/Slave relationship. CUCM controls it.

37. What is Mobile Voice Access?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Re-enter the mgcp bind control and mgcp bind media commands.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

38. What is Phone NTP Reference used for?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
To notify SIP Phones of NTP
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)
Debug ephone moh

39. How do you allow the Calling Name to be sent to the PSTN on a router?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
#voice service voip| #no supplementary-service h225-notify cid-update
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
When they are explicitly matched in a destination-pattern in a dial-peer.

40. How do you set up redundancy on outbound dial-peers on an H323 gateway?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
(config)#sh cdp neigh detail
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.

41. What does an H323 GW require that MGCP GWs do not?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Dial-peers

42. In Gatekeeper CAC how do you restrict a specific endpoint?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
#voice service voip| #no supplementary-service h225-notify cid-update

43. 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
SERVICE PARAMETER> Automated Alternate Routing Enable > True
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Re-enter the mgcp bind control and mgcp bind media commands.

44. What types of digit manipulation can you perform at the route pattern?
Single Number Reach (Mobile Connect)
Show ccm-manager
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
voice register pool|and|voice register dn

45. Is the order of the MRGs in an MRGL significant?
SIP Route Pattern over a SIP Trunk.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

46. On an H323 GW - how do you adjust the timers for redundancy hunting?
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
(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
Put them in a route list.
Reroute when here is WAN congestion.

47. How do you verify that NTP is working on the CUCM server?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
At the CLI: utils ntp status
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

48. How do you allow H323 calls to be preserved should the primary H323 GW fail?
Bandwidth
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

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

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50. Is the + character supported on a VOIP dial-peer?

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