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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. Calls coming from CUCM to PSTN need what?
'Use before the ... so XXXX
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.
Outbound dial-peers
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.

2. On the RDP What is the Rerouting CSS used for?
Single Number Reach (Mobile Connect)
Device Pool Locations
Call-manager-fallback or telephony-service mode 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

3. What is the bit rate for a G.729 call excluding layer 2?
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
Ccm-manager music-on-hold
8kb/s
'Service Parameters --> Auto Answer Timer

4. Maximum Wait Time for Desk Pickup?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Debug isdn q931

5. How do you test a Voice Translation Rule?
#test voice translation rule 1 <input to test>
Ccm-manager music-on-hold
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
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 ....

6. Privacy is enabled system-wide in CUCM by default. (T or F)
1 ... this is not optional!
TRUE
The name configuration field in ephone-dn and voice register dn
(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

7. 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)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
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

8. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

9. What are the commands to configure SIP phones in CME?
Apply it to the ephone.
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
#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
(config)#sh cdp neigh detail

10. How do you allow the Calling Name to be sent to the PSTN on a router?
Outbound dial-peers
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Dial-peers
(config)#sh cdp neigh detail

11. What does an H323 GW require that MGCP GWs do not?
Dial-peers
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
#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
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

12. How do you configure an MGCP GW? (router side)
Top Down means that channel 1 will be the first channel used to place outgoing calls.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
(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

13. What are the commands to configure NTP in IOS?
'Service Parameters --> Auto Answer Timer
When they are explicitly matched in a destination-pattern in a dial-peer.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

14. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Service Parameters --> Mobile Voice Access Number

15. How would you verify that DHCP is working in IOS?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
(config)#sh cdp neigh detail
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

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

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17. '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
...
Max-dn|max-ephone|ip source-address
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

18. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
Reroute when there is a WAN Outage.
In the CUCM 7.0.1 Release Notes ... search for restore.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
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

19. 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?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Matches any length dialed number and truncates it to 4 digits.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Mobile Voice Access

20. How can you confirm the MGCP GW is registered to CUCM - in IOS?
Sh ccm-manager
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Ccm-manager music-on-hold
When they are explicitly matched in a destination-pattern in a dial-peer.

21. How do you configure phone ports on an ESW?
The Called/Calling Transformations are superceded.
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
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
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

22. How do you disable KPML?
Assign SIP Dial Rules
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Apply voice-class codec
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

23. What's the difference between AAR and SRST?
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
(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
It's best to strip digits at the voice port.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

24. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
Reroute when here is WAN congestion.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Call Simulator. You can use to validate path from router to the PSTN.
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

25. What is Mobile Voice Access?
Via the voice hunt-group parallel command
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Matches any length dialed number and truncates it to 4 digits.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

26. What does rule 1 /^.*(....$)/ /1/ do?
Matches any length dialed number and truncates it to 4 digits.
#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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
...

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
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
Sh ip rsvp reservation||sh sccp connections
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.

28. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
...
Mobile Connect
To notify SIP Phones of NTP

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

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30. What are the base telephony-service commands for CME?
...
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

31. Name 4 useful show commands for active calls.
(config)#sh cdp neigh detail
The name configuration field in ephone-dn and voice register dn
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

32. How do you configure a gateway to register with gatekeeper?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
#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
...

33. How do you enable AAR system wide?
In the CUCM 7.0.1 Release Notes ... search for restore.
#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
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

34. How is NTP sych setup in CUCM?
CUCM OS Administration Settings --> NTP Servers
...
Call Simulator. You can use to validate path from router to the PSTN.
Debug ephone moh

35. How do you allow H323 calls to be preserved should the primary H323 GW fail?
Mobile Connect
Bottom up - to prevent glare.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Debug isdn q931

36. What does the Transfer On-hook Enabled Service Parameter do?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
Max-dn|max-ephone|ip source-address
Allows you to transfer by only pressing the Transfer button once.

37. How do you setup AutoRegistration in CUCM?
#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
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
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

38. How do you enable Extension Mobility for a device in CME?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
8kb/s
Assign a logout-profile to the ephone.
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

39. Does CUCM support RSVP natively?
Via the voice hunt-group parallel command
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Phone number followed by domain name.|i.e. 3006@ipxcme.com

40. What commands are needed to configure the voice register pool in CME?
Bandwidth
Call Simulator. You can use to validate path from router to the PSTN.
Assign a logout-profile to the ephone.
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

41. What does Display-IE do?
Device Pool Locations
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Sends the Calling Name.
...

42. How do you set the inter-digit timeout for SIP phones in CME?
(config)#voice register dialplan
Sh ccm-manager
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.

43. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
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
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
Via the voice hunt-group parallel command

44. What do you need to do to activate the CME GUI?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Push as much digit manipulation to the router as possible - so you can re-use it for 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

45. Which takes precedence Device Locations or Device Pool Locations?
Device Pool Locations
(config)#sh cdp neigh detail
#test voice translation rule 1 <input to test>
SERVICE PARAMETER> Automated Alternate Routing Enable > True

46. On the RDP What is the CSS 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:(SUB IP)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #voice-clas
Apply it to the ephone.
voice translation-profile|and|voice translation-rule
Mobile Voice Access

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

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48. What are the basic ephone-dn and ephone commands?
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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
Max-dn|max-ephone|ip source-address
...

49. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
SIP Dial Rules
Call forwarding between voip to voip (when CUBE is in play)
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Debug isdn q931

50. How do you configure DHCP in IOS?
(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
Reroute when there is a WAN Outage.
To enable two-stage dialing.
Call forwarding between voip to voip (when CUBE is in play)