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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 configure a gateway to register with gatekeeper?
CUCM OS Administration Settings --> NTP Servers
(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
#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
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

2. How much bandwidth does a G.729 call including layer 3 require?
When they are explicitly matched in a destination-pattern in a dial-peer.
SIP Dial Rules
#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
24kb/s

3. 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
1 ... this is not optional!
SIP Route Pattern over a SIP Trunk.
...
Debug ephone moh

4. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
#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
Service Parameter --> Block OffNet To OffNet Transfer --> True
In the CUCM 7.0.1 Release Notes ... search for restore.

5. Which takes precedence Device Locations or Device Pool Locations?
Via the voice hunt-group parallel command
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
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Device Pool Locations

6. In CME Where is the Calling Name derived from?
Device Pool Locations
The name configuration field in ephone-dn and voice register dn
To notify SIP Phones of NTP
#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

7. What is the Service Parameter to allow the phone that is picking ip to auto answer?
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
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Auto Call Pickup Enabled
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie

8. In Gatekeeper CAC how do you restrict a specific endpoint?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Call forwarding between voip to voip (when CUBE is in play)
Assign SIP Dial Rules

9. How do you support multiple codecs on a dialpeer?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Apply voice-class codec
Debug isdn q931
#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

10. What terminology translates to SRST?
...
Reroute when there is a WAN Outage.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

11. How do you get around relying on DNS for your CUCMs?
Mobile Voice Access
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
In CUCM - configure CFUR to point to it's E164 number.

12. How do you configure a SIP Trunk? (router side)
(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
Via the voice hunt-group parallel command
Auto Call Pickup Enabled
Dial-peers

13. What is the bit rate for a G.729 call excluding layer 2?

8kb/s
It's best to strip digits at the voice port.
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

14. What must you do for a BACD script to work on a CME router?
#test voice translation rule 1 <input to test>
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
It's best to strip digits at the voice port.

15. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
#after-hours block pattern
Service Parameters --> Mobile Voice Access Number
Re-enter the mgcp bind control and mgcp bind media commands.
Mobile Voice Access

16. What CUCM services should you activate?
#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
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Call forwarding between voip to voip (when CUBE is in play)

17. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
Auto Call Pickup Enabled
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
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

18. 'How do you inform a SIP phone of NTP information?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
...
Via the voice hunt-group parallel command

19. Does CUCM support RSVP natively?
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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Max-dn|max-ephone|ip source-address
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.

20. On an MGCP GW - how could you see the primary and backup CUCM servers?
8kb/s
Show ccm-manager
Debug ephone moh
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

21. 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?
Master/Slave relationship. CUCM controls it.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
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)
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

22. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Bottom up - to prevent glare.
#voice service voip| #no supplementary-service h225-notify cid-update
Precede the # with a > ... so 9011*>#
In CUCM - configure CFUR to point to it's E164 number.

23. How can you confirm the MGCP GW is registered to CUCM - in IOS?
Assign SIP Dial Rules
At the CLI: utils ntp status
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
Sh ccm-manager

24. What are the 3 mandatory commands within call-manager-fallback?
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 .)
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Max-dn|max-ephone|ip source-address
(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

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

26. What are the commands to create vlans on an ESW?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
#voice service voip| #no supplementary-service h225-notify cid-update

27. Two useful troubleshooting commands for CUCME?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Sends the Calling Name.
Bottom up - to prevent glare.
sh ephone reg|and|debug voip dialpeer

28. How do you enable AAR system wide?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Sh ip rsvp reservation||sh sccp connections
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

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

30. What are the steps to configure Single Number Reach and Mobile Voice Access?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Debug isdn q931
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

31. When can you not use a Standard Local Route Group?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Re-enter the mgcp bind control and mgcp bind media commands.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
When configuring TEHO.

32. When setting up SIP URI where do you configure the CUCM's domain name?'
Apply voice-class codec
Reroute when there is a WAN Outage.
(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
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name

33. What types of digit manipulation can you perform at the route pattern?
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.
Sh ip rsvp reservation||sh sccp connections
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
SIP Route Pattern over a SIP Trunk.

34. How do you set the inter-digit timeout for SIP phones in CME?
8kb/s
(config)#voice register dialplan
At the CLI: utils ntp status

35. What are the commands to set up the PRI?

36. Calls coming from CUCM to PSTN need what?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Outbound dial-peers
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

37. What are the basic SCCP Commands fro telephony-service in CUCME?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Service Parameters --> Mobile Voice Access Number

38. How do you configure an MGCP GW? (router side)
(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
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.
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

39. What are the commands to configure an H323 GW?
voice translation-profile|and|voice translation-rule
#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
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.

40. What dtmf-relay type do you use for an H323 GW?
...
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Dtmf-relay h245-alpha
(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

41. '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
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)
Max-dn|max-ephone|ip source-address
1 ... this is not optional!
...

42. How do you ensure that G.711 only is used?
Single Number Reach (Mobile Connect)
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
#voice service voip| #no supplementary-service h225-notify cid-update
Put them in a route list.

43. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
To notify SIP Phones of NTP
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au

44. How do you configure CUCM redundancy on an H323 gateway?

(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
#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)#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

45. What are the commands to manually configure an MGCP gateway?
(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 --> Block OffNet To OffNet Transfer --> True
#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
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.

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

47. How do you create a trunk on the router side?
(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
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
#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
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

48. What does the Transfer On-hook Enabled Service Parameter do?
(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
Allows you to transfer by only pressing the Transfer button once.
It's best to strip digits at the voice port.
The Called/Calling Transformations are superceded.

49. What is the Enable Enterprise Feature Access Service Parameter used for?
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
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
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
To enable two-stage dialing.

50. When do you use translate called? Translate calling?
Automatically configures the MGCP GW for you.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Top Down means that channel 1 will be the first channel used to place outgoing calls.
When they are explicitly matched in a destination-pattern in a dial-peer.