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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. When do you use translate called? Translate calling?
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)#sh cdp neigh detail
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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

2. What is SIP URI?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
When multiple sites use the same route pattern - do your digit manipulation on a route list.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Dtmf-relay h245-alpha

3. How do you get around relying on DNS for your CUCMs?
24kb/s
Debug ephone moh
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
(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

4. What are the basic SCCP Commands fro telephony-service in CUCME?
IP Voice Media Streaming App
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Mobile Connect
Show ccm-manager

5. What is CSIM?
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 .)
Sends the Calling Name.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Call Simulator. You can use to validate path from router to the PSTN.

6. How do you allow the Calling Name to be sent to the PSTN on a router?
CUCM OS Administration Settings --> NTP Servers

Call forwarding between voip to voip (when CUBE is in play)
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie

7. How do you configure a SIP Trunk? (router side)
(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
At the CLI: utils ntp status
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

8. How do you disable KPML?
(config)#sh cdp neigh detail
Assign SIP Dial Rules
Master/Slave relationship. CUCM controls it.
To notify SIP Phones of NTP

9. How do you configure SRST?
To enable two-stage dialing.
Call-manager-fallback or telephony-service mode SRST
sh ephone reg|and|debug voip dialpeer
Bandwidth

10. What terminology translates to AAR?
Reroute when here is WAN congestion.
Higher
Allows you to transfer by only pressing the Transfer button once.
(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

11. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
Dial-peers
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.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
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.

12. 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
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Precede the # with a > ... so 9011*>#

13. 'If you're not getting a DHCP address from CUCM what then?
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
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Debug ephone moh

14. Describe the relationship between route patterns and end devices.
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.
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.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Service Parameters --> Mobile Voice Access Number

15. When do you do digit manipulation at the route pattern as opposed to the route list?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
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
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.

16. What are the steps to integrate CUPS with CUCM?
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
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Apply voice-class codec
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

17. What does the Transfer On-hook Enabled Service Parameter do?
Allows you to transfer by only pressing the Transfer button once.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

18. How do you verify that NTP is working on the CUCM server?
At the CLI: utils ntp status
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
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
(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

19. What is Phone NTP Reference used for?
To notify SIP Phones of NTP
Sends the Calling Name.
Max-dn|max-ephone|ip source-address
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?
...
When they are explicitly matched in a destination-pattern in a dial-peer.
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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

21. What terminology translates to SRST?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
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.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Reroute when there is a WAN Outage.

22. How do you allow H323 calls to be preserved should the primary H323 GW fail?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Bandwidth
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

23. What does KPML do?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|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
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Configure it on the route group through the route list - then it will be local to the route list.

24. 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
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
Dtmf-relay h245-alpha

25. 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
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
8kb/s

26. What are the commands to configure SIP phones in CME?
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
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
1 ... this is not optional!
Bandwidth

27. What does an H323 GW require that MGCP GWs do not?
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
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
Dial-peers

28. How do you ensure that G.711 only is used?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Call Simulator. You can use to validate path from router to the PSTN.
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|(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

29. How do you configure an MGCP GW? (router side)
Mobile Voice Access
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
(config)#voice register dialplan
(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

30. How do you use the # as a string terminator within a SIP Dial Rule?
When they are explicitly matched in a destination-pattern in a dial-peer.
Precede the # with a > ... so 9011*>#
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

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

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32. How do you prevent toll fraud on CUCM?
Precede the # with a > ... so 9011*>#
voice translation-profile|and|voice translation-rule
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Auto Call Pickup Enabled

33. What kind of relationship does an MGCP gateway have with CUCM?
Master/Slave relationship. CUCM controls it.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.

Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

34. What is the Enable Enterprise Feature Access Service Parameter used for?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
To enable two-stage dialing.
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
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)

35. What are the commands to configure NTP in IOS?
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
Assign SIP Dial Rules
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Allows you to transfer by only pressing the Transfer button once.

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

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37. Calls coming from CUCM to PSTN need what?
#voice service voip| #no supplementary-service h225-notify cid-update
Top Down means that channel 1 will be the first channel used to place outgoing calls.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Outbound dial-peers

38. What are the 3 mandatory commands within call-manager-fallback?
Dtmf-relay h245-alpha
Assign a logout-profile to the ephone.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Max-dn|max-ephone|ip source-address

39. 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?
Call forwarding between voip to voip (when CUBE is in play)
Bottom up - to prevent glare.
Automatically configures the MGCP GW for you.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.

40. 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!
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
Single Number Reach (Mobile Connect)
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

41. How do you block an external call from being transferred back out to the pstn by an internal user?
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.
Service Parameter --> Block OffNet To OffNet Transfer --> True
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Automatically configures the MGCP GW for you.

42. 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?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
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.
Outbound dial-peers
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

43. How do you change modes in voice register global?
Service Parameter --> Block OffNet To OffNet Transfer --> True
IP Voice Media Streaming App
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

44. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
Dtmf-relay h245-alpha
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
8kb/s

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

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46. How do you support multiple codecs on a dialpeer?
Apply voice-class codec
Master/Slave relationship. CUCM controls it.
Configure it on the route group through the route list - then it will be local to the route list.
#voice service voip| #no supplementary-service h225-notify cid-update

47. What would force you to use telephony-service to configure SRST?
IP Voice Media Streaming App
#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
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
#voice service voip| #no supplementary-service h225-notify cid-update

48. 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 os truly unsurpassed. I have opened up a wealth of opportunities for myself and my family and
Max-dn|max-ephone|ip source-address

Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
...

49. On the RDP What is the CSS used for?
Mobile Voice Access
(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
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

50. How do you use an ephone template?
Assign SIP Dial Rules
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
8kb/s
Apply it to the ephone.