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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 verify that NTP is working on the CUCM server?
Max-dn|max-ephone|ip source-address
Sends the Calling Name.
At the CLI: utils ntp status

2. '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
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Service Parameter --> Block OffNet To OffNet Transfer --> True

3. Does CUCM support RSVP natively?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
When they are explicitly matched in a destination-pattern in a dial-peer.
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

4. How do you configure AAR?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Allows you to transfer by only pressing the Transfer button once.

5. What are the basic ephone-dn and ephone commands?
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
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
Higher
'Use before the ... so XXXX

6. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the 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
Sends the Calling Name.
Mobile Connect
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

7. How do you configure class of service (CoS) in CUCM? CME?
Sh ip rsvp reservation||sh sccp connections
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
'Service Parameters --> Auto Answer Timer
The Called/Calling Transformations are superceded.

8. How do you use an ephone template?
Bandwidth
...
Via the voice hunt-group parallel command
Apply it to the ephone.

9. Which takes precedence Device Locations or Device Pool Locations?
(config)#voice register dialplan
Device Pool Locations
24kb/s
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

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

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11. How do you configure an MGCP GW? (router side)
#voice service voip| #no supplementary-service h225-notify cid-update
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
...
(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

12. What CUCM services should you activate?
SIP Route Pattern over a SIP Trunk.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Via the voice hunt-group parallel command

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

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14. What is the Enable Enterprise Feature Access Service Parameter used for?
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
Outbound dial-peers
To enable two-stage dialing.
Apply it to the ephone.

15. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
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.
On CUCM it's identical to adding an H323 GW.

16. When using drop-through-option What is the max number of huntgroups?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Apply voice-class codec
On CUCM it's identical to adding an H323 GW.
1 ... this is not optional!

17. What does Display-IE do?
The Called/Calling Transformations are superceded.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
When configuring TEHO.
Sends the Calling Name.

18. What is the full E164 format?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
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 .)
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
SIP Route Pattern over a SIP Trunk.

19. On an H323 GW - how do you adjust the timers for redundancy hunting?
Dtmf-relay h245-alpha
Re-enter the mgcp bind control and mgcp bind media commands.
Service Parameters --> Mobile Voice Access Number
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

20. In CME Where is the Calling Name derived from?
Master/Slave relationship. CUCM controls it.
Sends the Calling Name.
The name configuration field in ephone-dn and voice register dn
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

21. How do you test a Voice Translation Rule?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Sh ccm-manager
#test voice translation rule 1 <input to test>

22. How do you setup AutoRegistration in CUCM?
Outbound dial-peers
sh ephone reg|and|debug voip dialpeer
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

23. What's the best way to perform digit manipulation on a route group?
When configuring TEHO.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Apply voice-class codec
Configure it on the route group through the route list - then it will be local to the route list.

24. How do you support multiple codecs on a dialpeer?
#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
Apply voice-class codec
Matches any length dialed number and truncates it to 4 digits.
Show ccm-manager

25. On the RDP What is the CSS used for?
#voice service voip| #no supplementary-service h225-notify cid-update
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Mobile Voice Access
Apply voice-class codec

26. When a CUCM device dials a number - what happens?
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 .)
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.
(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
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

27. What are the commands to manually configure an MGCP gateway?
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
#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
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie

28. What are the commands to configure SRST in fallback?
Automatically configures the MGCP GW for you.
Sh ip rsvp reservation||sh sccp connections
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

29. What are the commands to configure SIP phones in CME?
Single Number Reach (Mobile Connect)
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
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
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.

30. What does rule 1 /^.*(....$)/ /1/ do?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Matches any length dialed number and truncates it to 4 digits.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Assign SIP Dial Rules

31. What would force you to use telephony-service to configure SRST?
Service Parameters --> Mobile Voice Access Number
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Call Simulator. You can use to validate path from router to the PSTN.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

32. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Sh ccm-manager
Apply it to the ephone.
voice register pool|and|voice register dn

33. How do you ensure that G.711 only is used?
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
To notify SIP Phones of NTP

34. How do you prevent H323 caller-id updates to CUCM
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.
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
#voice service voip| #no supplementary-service h225-notify cid-update
To enable two-stage dialing.

35. When a device is roaming to a different site How does it get the Roaming Sensitive Settings from the Device Pool?
Put them in a route list.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
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.

36. When can you not use a Standard Local Route Group?
#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
Master/Slave relationship. CUCM controls it.
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 configuring TEHO.

37. When setting up SIP URI where do you configure the CUCM's domain name?'
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
To enable two-stage dialing.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.

38. What is Phone NTP Reference used for?
#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
Sh ccm-manager
Auto Call Pickup Enabled
To notify SIP Phones of NTP

39. What are the commands to configure an H323 GW?
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
#after-hours block pattern
#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
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'

40. What is the Service Parameter to allow the phone that is picking ip to auto answer?
Auto Call Pickup Enabled
sh ephone reg|and|debug voip dialpeer
#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

41. What types of digit manipulation can you perform at the route pattern?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
At the CLI: utils ntp status
In the CUCM 7.0.1 Release Notes ... search for restore.
SIP Route Pattern over a SIP Trunk.

42. If you're using Called/Calling Party Transformations at the Route Pattern and tranformations at the route list ... What does the RP affect and What does the RL affect?
Automatically configures the MGCP GW for you.
(config)#voice register dialplan
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

43. How do you configure CUCM redundancy on an H323 gateway?
#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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
(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
Device Pool Locations

44. What does the Transfer On-hook Enabled Service Parameter do?
CUCM OS Administration Settings --> NTP Servers
Allows you to transfer by only pressing the Transfer button once.
Mobile Connect
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

45. How do you enable AAR system wide?
#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
Ccm-manager music-on-hold
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.

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

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47. How do you create a trunk on the switch side?
#voice service voip| #no supplementary-service h225-notify cid-update
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

48. How do you allow H323 calls to be preserved should the primary H323 GW fail?
24kb/s
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Reroute when here is WAN congestion.

49. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Mobile Voice Access
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.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

50. What is the bit rate for a G.729 call excluding layer 2?
8kb/s
IP Voice Media Streaming App
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
To notify SIP Phones of NTP