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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 enable AAR?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Device Pool Locations
8kb/s
SERVICE PARAMETER> Automated Alternate Routing Enable > True

2. How would you enable security on a GK?
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
Configure it on the route group through the route list - then it will be local to the route list.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
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 ....

3. Name 4 useful show commands for active calls.
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
Master/Slave relationship. CUCM controls it.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
(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

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

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5. When do you use translate called? Translate calling?
Bandwidth
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Precede the # with a > ... so 9011*>#
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

6. What are the commands to manually configure an MGCP gateway?
#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
8kb/s
Outbound dial-peers
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

7. How do you configure class of service (CoS) in CUCM? CME?
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.
Dial-peers
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

8. How do you verify that NTP is working on the CUCM server?
8kb/s
At the CLI: utils ntp status
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
TRUE

9. What's the difference between AAR and SRST?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Show ccm-manager
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

10. 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?
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
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
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.
Reroute when here is WAN congestion.

11. How do you enable Extension Mobility for a device in CME?
Reroute when here is WAN congestion.
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
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Assign a logout-profile to the ephone.

12. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
When multiple sites use the same route pattern - do your digit manipulation on a route list.

13. What are the commands to configure an H323 GW?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
#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
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
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)

14. What are the commands to configure SRST in fallback?
Configure it on the route group through the route list - then it will be local to the route list.
Debug isdn q931
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

15. 'If you want to make changes to any softkeys where do you do it?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
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
Single Number Reach (Mobile Connect)

16. How do you configure an MGCP GW? (router side)
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Call Simulator. You can use to validate path from router to the PSTN.
(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

17. How do you configure CUBE?

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18. What is CSIM?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Call Simulator. You can use to validate path from router to the PSTN.
8kb/s
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

19. On an H323 GW - how do you adjust the timers for redundancy hunting?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
#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

h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

20. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
Debug isdn q931
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!

21. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
(config)#voice register dialplan
SIP Dial Rules
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

22. Does CUCM support RSVP natively?
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.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
sh ephone reg|and|debug voip dialpeer
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

23. What should you configure before entering auto qos?
Reroute when there is a WAN Outage.
Bandwidth
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
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

24. What are the commands to configure a SIP phone in CUCME?
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
In CUCM - configure CFUR to point to it's E164 number.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
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

25. When using drop-through-option What is the max number of huntgroups?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
24kb/s
1 ... this is not optional!
It's best to strip digits at the voice port.

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

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27. What CUCM services should you activate?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
(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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Mobile Voice Access

28. When dealing with an MGCP GW there is no intelligence on the router. How would you strip an incoming Called Number from 10 to 4 digits?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Re-enter the mgcp bind control and mgcp bind media commands.

29. Two useful troubleshooting commands for CUCME?
8kb/s
Debug isdn q931
...
sh ephone reg|and|debug voip dialpeer

30. How do you configure AAR?
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 ....
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
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

31. 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!
Bandwidth
(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
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

32. Where can you assign the AAR Group?
Sends the Calling Name.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Dial-peers

33. When can you not use a Standard Local Route Group?
Dial-peers
When configuring TEHO.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
#after-hours block pattern

34. 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.
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)#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
In CUCM - configure CFUR to point to it's E164 number.

35. How do you block an external call from being transferred back out to the pstn by an internal user?
Service Parameter --> Block OffNet To OffNet Transfer --> True
(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
Bandwidth
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

36. What is call-forward pattern used for?
#after-hours block pattern
Assign SIP Dial Rules
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.
Call forwarding between voip to voip (when CUBE is in play)

37. What are the commands to configure SIP phones in CME?
#after-hours block pattern
In the CUCM 7.0.1 Release Notes ... search for restore.
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
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

38. How much bandwidth does a G.729 call including layer 3 require?
24kb/s
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
To notify SIP Phones of NTP
(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

39. What terminology translates to SRST?
#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
Assign a logout-profile to the ephone.
Reroute when there is a WAN Outage.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable

40. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
SIP Route Pattern over a SIP Trunk.
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

41. On the RDP What is the Rerouting CSS used for?
Single Number Reach (Mobile Connect)
The name configuration field in ephone-dn and voice register dn
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
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.

42. What kind of relationship does an MGCP gateway have with CUCM?
Master/Slave relationship. CUCM controls it.

...
(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

43. What are the 3 mandatory commands within call-manager-fallback?
8kb/s
Max-dn|max-ephone|ip source-address
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
#voice service voip| #no supplementary-service h225-notify cid-update

44. How do do configure TEHO?
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
Service Parameters --> Mobile Voice Access Number
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
To notify SIP Phones of NTP

45. How do you change modes in voice register global?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Service Parameters --> Mobile Voice Access Number

46. How do you use an ephone template?
Service Parameters --> Mobile Voice Access Number
Apply it to the ephone.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
#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

47. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
24kb/s
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
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
#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

48. Describe the relationship between route patterns and end devices.
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.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
'Service Parameters --> Auto Answer Timer
Max-dn|max-ephone|ip source-address

49. How do you configure a SIP Trunk? (router side)
To notify SIP Phones of NTP
SIP Dial Rules
(config)#sh cdp neigh detail
(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

50. How do you use the # as a string terminator within a SIP Dial Rule?
...
Precede the # with a > ... so 9011*>#
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.