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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 block an external call from being transferred back out to the pstn by an internal user?
Max-dn|max-ephone|ip source-address
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Master/Slave relationship. CUCM controls it.
Service Parameter --> Block OffNet To OffNet Transfer --> True

2. What types of digit manipulation can you perform at the route pattern?
Auto Call Pickup Enabled
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Phone number followed by domain name.|i.e. 3006@ipxcme.com
CUCM OS Administration Settings --> NTP Servers

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

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4. Calls coming from CUCM to PSTN need what?

No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Outbound dial-peers
Device Pool Locations

5. What is call-forward pattern used for?
Call forwarding between voip to voip (when CUBE is in play)
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Sends every digit to CUCM in real-time with digit-analysis being handled by 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:(SUB IP)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #voice-clas

6. How do you enable AAR?
In the CUCM 7.0.1 Release Notes ... search for restore.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Apply voice-class codec

7. How do you test a Voice Translation Rule?
Master/Slave relationship. CUCM controls it.
#test voice translation rule 1 <input to test>
Debug ephone moh
Re-enter the mgcp bind control and mgcp bind media commands.

8. What are the commands to create vlans on an ESW?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
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
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Show ccm-manager

9. How do you prevent toll fraud on CUCM?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
voice register pool|and|voice register dn
Reroute when here is WAN congestion.

10. What terminology translates to SRST?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
IP Voice Media Streaming App
Reroute when there is a WAN Outage.
Dtmf-relay h245-alpha

11. What is Mobile Voice Access?
Matches any length dialed number and truncates it to 4 digits.
#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
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
SERVICE PARAMETER> Automated Alternate Routing Enable > True

12. On the RDP What is the CSS used for?
Mobile Voice Access
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Show ccm-manager
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

13. When do you use translate called? Translate calling?
Assign SIP Dial Rules
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
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

14. What is the Enable Enterprise Feature Access Service Parameter used for?
To enable two-stage dialing.
To notify SIP Phones of NTP
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Outbound dial-peers

15. What are the commands to configure an H323 GW?
On CUCM it's identical to adding an H323 GW.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
#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
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

16. How can you confirm the MGCP GW is registered to CUCM - in IOS?
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
...
Sh ccm-manager
Configure it on the route group through the route list - then it will be local to the route list.

17. What does ccm-manager config server [IP]</ip> do?
Outbound dial-peers
The Called/Calling Transformations are superceded.
Automatically configures the MGCP GW for you.
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.

18. When are digits stripped in a gateway?
When they are explicitly matched in a destination-pattern in a dial-peer.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

19. When setting up SIP URI where do you configure the CUCM's domain name?'
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name

Master/Slave relationship. CUCM controls it.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

20. How do you configure class of service (CoS) in CUCM? CME?
Call Simulator. You can use to validate path from router to the PSTN.
When they are explicitly matched in a destination-pattern in a dial-peer.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
#test voice translation rule 1 <input to test>

21. How do you configure a SIP Trunk? (router side)
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.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
(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
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

22. 'How do you inform a SIP phone of NTP information?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
sh ephone reg|and|debug voip dialpeer
'Use before the ... so XXXX
#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

23. 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 .)
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 ....
(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
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

24. 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
Allows you to transfer by only pressing the Transfer button once.
Master/Slave relationship. CUCM controls it.
It's best to strip digits at the voice port.

25. How do you prioritize route groups?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
#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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Put them in a route list.

26. What would force you to use telephony-service to configure SRST?
Precede the # with a > ... so 9011*>#
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Top Down means that channel 1 will be the first channel used to place outgoing calls.

27. 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?
IP Voice Media Streaming App
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)
Top Down means that channel 1 will be the first channel used to place outgoing calls.
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

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?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Automatically configures the MGCP GW for you.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

29. How do you configure AAR?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Dial-peers
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 .)
Sh ccm-manager

30. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?

The Called/Calling Transformations are superceded.
Show ccm-manager
(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

31. What does Display-IE do?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Sends the Calling Name.
Call Simulator. You can use to validate path from router to the PSTN.

32. How do you enable AAR system wide?
Re-enter the mgcp bind control and mgcp bind media commands.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Call forwarding between voip to voip (when CUBE is in play)
CUCM Service Parameters --> Automated Alternate Routing Enable --> True

33. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
'Use before the ... so XXXX
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
The Called/Calling Transformations are superceded.

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

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35. What does an H323 GW require that MGCP GWs do not?
#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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Dial-peers
(config)#sh cdp neigh detail

36. How do you prevent H323 caller-id updates to CUCM
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.
#voice service voip| #no supplementary-service h225-notify cid-update
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

37. On an H323 GW - how do you adjust the timers for redundancy hunting?
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
Via the voice hunt-group parallel command
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Show ccm-manager

38. What commands are needed to configure the voice register pool in CME?
#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
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
Reroute when here is WAN congestion.

39. What are the commands to configure SRST in fallback?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Matches any length dialed number and truncates it to 4 digits.
Precede the # with a > ... so 9011*>#
#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

40. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
Single Number Reach (Mobile Connect)
Re-enter the mgcp bind control and mgcp bind media commands.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

41. How do do configure TEHO?
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
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

Call forwarding between voip to voip (when CUBE is in play)

42. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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
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.
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 .)
Debug isdn q931

43. Privacy is enabled system-wide in CUCM by default. (T or F)
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
#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
TRUE
CUCM Service Parameters --> Call Park Reversion Timer

44. In CME Where is the Calling Name derived from?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
The name configuration field in ephone-dn and voice register dn
Debug ephone moh

45. What are the commands to create the L3 routing interface for VLANS (SVI)?
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
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
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Apply voice-class codec

46. What are the commands to configure a SIP phone in CUCME?
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
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.
IP Voice Media Streaming App
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

47. Which takes precedence Device Locations or Device Pool Locations?
Device Pool Locations
Via the voice hunt-group parallel command
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.
Auto Call Pickup Enabled

48. How do you use the # as a string terminator within a SIP Dial Rule?
Precede the # with a > ... so 9011*>#
It's best to strip digits at the voice port.
Bottom up - to prevent glare.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

49. What kind of relationship does an MGCP gateway have with CUCM?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Assign SIP Dial Rules
Master/Slave relationship. CUCM controls it.
voice translation-profile|and|voice translation-rule

50. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?

Bottom up - to prevent glare.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Sh ccm-manager