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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. With the gw-priority command - does higher or lower priority take precedence?
#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
Higher
Apply voice-class codec
Phone number followed by domain name.|i.e. 3006@ipxcme.com

2. What does the Transfer On-hook Enabled Service Parameter do?
#voice service voip| #no supplementary-service h225-notify cid-update
Sends the Calling Name.
Allows you to transfer by only pressing the Transfer button once.
Top Down means that channel 1 will be the first channel used to place outgoing calls.

3. Name 4 useful show commands for active calls.
voice register pool|and|voice register dn
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
(config)#sh cdp neigh detail

4. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
#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
The Called/Calling Transformations are superceded.
#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|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

5. What does ccm-manager config server [IP]</ip> do?
Automatically configures the MGCP GW for you.
#voice service voip| #no supplementary-service h225-notify cid-update
Call forwarding between voip to voip (when CUBE is in play)
Service Parameters --> Mobile Voice Access Number

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

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7. 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
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

8. What is a best practice for digit manipulation - in regards to H323 GWs?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
It's best to strip digits at the voice port.
8kb/s
TRUE

9. 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?

#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
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.

10. Describe how you configure SIP URI functionality.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

Service Parameter --> Block OffNet To OffNet Transfer --> True
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!

11. What is the Enable Enterprise Feature Access Service Parameter used for?
...
To enable two-stage dialing.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Reroute when here is WAN congestion.

12. How do you block calls under call-manager-fallback?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
#after-hours block pattern
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
In the CUCM 7.0.1 Release Notes ... search for restore.

13. 'If you want to make changes to any softkeys where do you do it?
Single Number Reach (Mobile Connect)
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.

14. What are the commands to configure an H323 GW?
Bottom up - to prevent glare.
Sh ip rsvp reservation||sh sccp connections
#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
Mobile Connect

15. Maximum Wait Time for Desk Pickup?
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
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
voice register pool|and|voice register dn
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.

16. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
voice register pool|and|voice register dn
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

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

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18. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Precede the # with a > ... so 9011*>#
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

19. What CUCM services should you activate?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable

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
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.

20. How do you set the Call Park Reversion Timer?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
CUCM Service Parameters --> Call Park Reversion Timer
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
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.

21. What are the 3 mandatory commands within call-manager-fallback?
Max-dn|max-ephone|ip source-address
Via the voice hunt-group parallel command
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.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

22. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
Assign SIP Dial Rules
Top Down means that channel 1 will be the first channel used to place outgoing calls.
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Reroute when here is WAN congestion.

23. How do you use the # as a string terminator within a SIP Dial Rule?
Precede the # with a > ... so 9011*>#
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Assign a logout-profile to the ephone.
...

24. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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.
Sh ccm-manager
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 ....
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

25. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
CUCM OS Administration Settings --> NTP Servers
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.
Via the voice hunt-group parallel command
Re-enter the mgcp bind control and mgcp bind media commands.

26. How do you create a trunk on the switch side?
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Sends the Calling Name.
#after-hours block pattern
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 .)

27. What should you configure before entering auto qos?
Assign SIP Dial Rules
(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
#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
Bandwidth

28. 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
...
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Matches any length dialed number and truncates it to 4 digits.
Mobile Voice Access

29. How do you enable Extension Mobility for a device in CME?
Assign a logout-profile to the ephone.
(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
Sh ip rsvp reservation||sh sccp connections
TRUE

30. How do you setup AutoRegistration in CUCM?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.

31. What are the commands to configure a T1/E1 PRI?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Apply voice-class codec
(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

32. What is the bit rate for a G.729 call excluding layer 2?
8kb/s
Max-dn|max-ephone|ip source-address
(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
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

33. 'How do you inform a SIP phone of NTP information?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
CUCM OS Administration Settings --> NTP Servers
Apply voice-class codec
Top Down means that channel 1 will be the first channel used to place outgoing calls.

34. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
SIP Dial Rules
SIP Route Pattern over a SIP Trunk.

35. What are the commands to create vlans on an ESW?
(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
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Automatically configures the MGCP GW for you.

36. In CME Where is the Calling Name derived from?

SERVICE PARAMETER> Automated Alternate Routing Enable > True
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
The name configuration field in ephone-dn and voice register dn

37. How do do configure TEHO?
Apply voice-class codec
(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
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
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

38. How do you prevent toll fraud on CME?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
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 ....
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

39. How do you verify that NTP is working on the CUCM server?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
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
At the CLI: utils ntp status

40. What would force you to use telephony-service to configure 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
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files

41. How do you prioritize route groups?

Put them in a route list.
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-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

42. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
#voice service voip| #no supplementary-service h225-notify cid-update
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

43. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
In the CUCM 7.0.1 Release Notes ... search for restore.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
To notify SIP Phones of NTP

44. What are the base telephony-service commands for CME?
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

45. Calls coming from CUCM to PSTN need what?
Matches any length dialed number and truncates it to 4 digits.
Outbound dial-peers
...
Redundant dial plan. One on CUCM and one on the routers via dial-peers.

46. How do you change modes in voice register global?
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.
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
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
Assign a logout-profile to the ephone.

47. What are the basic ephone-dn and ephone commands?
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.
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
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

48. How do you use an ephone template?
Reroute when there is a WAN Outage.
Apply it to the ephone.
Service Parameters --> Mobile Voice Access Number
...

49. What do you need to do to activate the CME GUI?
On CUCM it's identical to adding an H323 GW.
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
(config)#voice register dialplan
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

50. When do you do digit manipulation at the route pattern as opposed to the route list?
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
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Call Simulator. You can use to validate path from router to the PSTN.