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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 configure class of service (CoS) in CUCM? CME?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
#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
Call-manager-fallback or telephony-service mode SRST
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

2. Name 2 commands to verify RSVP 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)
Sends every digit to CUCM in real-time with digit-analysis being handled by 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
Sh ip rsvp reservation||sh sccp connections

3. What does KPML do?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
The name configuration field in ephone-dn and voice register dn
#after-hours block pattern
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

4. How do you configure DHCP in IOS?
When they are explicitly matched in a destination-pattern in a dial-peer.
(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
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
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.

5. How do you allow the Calling Name to be sent to the PSTN on a router?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Service Parameter --> Block OffNet To OffNet Transfer --> True
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

6. How do you configure phone ports on a 3750?
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
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
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

7. How would you enable security on a GK?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
#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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

8. What are the commands to create the L3 routing interface for VLANS (SVI)?
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
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
sh ephone reg|and|debug voip dialpeer
24kb/s

9. How do you see multicast packets being sent?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
In CUCM - configure CFUR to point to it's E164 number.
Debug ephone moh
Assign SIP Dial Rules

10. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
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 .)
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
Via the voice hunt-group parallel command

11. How do you create a trunk on the router side?
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
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
Reroute when here is WAN congestion.
(config)#voice register dialplan

12. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
voice translation-profile|and|voice translation-rule
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Call forwarding between voip to voip (when CUBE is in play)

13. How do you prevent toll fraud on CUCM?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
(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
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
To enable two-stage dialing.

14. 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
The Called/Calling Transformations are superceded.
...
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
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

15. When using drop-through-option What is the max number of huntgroups?
1 ... this is not optional!

Bandwidth
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.

16. What's the difference between AAR and SRST?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Debug ephone moh
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

17. How do you use the # as a string terminator within a SIP Dial Rule?
Precede the # with a > ... so 9011*>#
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
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
Apply voice-class codec

18. How do you configure a SIP Trunk? (router side)
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
(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
The name configuration field in ephone-dn and voice register dn
Assign SIP Dial Rules

19. How do you set the inter-digit timeout for SIP phones in CME?
(config)#voice register dialplan
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
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.

20. How do you set the Call Park Reversion Timer?
CUCM Service Parameters --> Call Park Reversion Timer
TRUE
SIP Dial Rules
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

21. How do you use an ephone template?
At the CLI: utils ntp status
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Reroute when there is a WAN Outage.
Apply it to the ephone.

22. 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?
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.
Higher
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
Call-manager-fallback or telephony-service mode SRST

23. What are the commands to configure a SIP phone in CUCME?
Dtmf-relay h245-alpha
TRUE
(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

24. What are the commands to manually configure an MGCP gateway?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Bottom up - to prevent glare.
#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
SIP Route Pattern over a SIP Trunk.

25. What CUCM services should you activate?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Apply voice-class codec
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name

26. What are the base telephony-service commands for CME?
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

27. How do you get around relying on DNS for your CUCMs?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
SIP Dial Rules
Put them in a route list.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

28. When a CUCM device dials a number - what happens?
voice translation-profile|and|voice translation-rule
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
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 .)
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

29. Which takes precedence Device Locations or Device Pool Locations?

Outbound dial-peers
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
Device Pool Locations

30. What are the 3 mandatory commands within call-manager-fallback?
Precede the # with a > ... so 9011*>#
Max-dn|max-ephone|ip source-address
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. Is the order of the MRGs in an MRGL significant?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Sh ccm-manager
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
sh ephone reg|and|debug voip dialpeer

32. How do you ensure that G.711 only is used?
CUCM OS Administration Settings --> NTP Servers
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Debug isdn q931
To enable two-stage dialing.

33. What is the Enable Enterprise Feature Access Service Parameter used for?
Sends the Calling Name.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
To enable two-stage dialing.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

34. When a device is roaming to a different site How does it get the Roaming Sensitive Settings from the Device Pool?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
sh ephone reg|and|debug voip dialpeer

35. What are the commands to configure NTP in IOS?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
When configuring TEHO.
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)
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

36. What is a best practice for digit manipulation - in regards to H323 GWs?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Sends the Calling Name.
Call forwarding between voip to voip (when CUBE is in play)
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

37. What does an H323 GW require that MGCP GWs do not?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Dial-peers
SIP Dial Rules
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name

38. How do you configure a gateway to register with gatekeeper?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Allows you to transfer by only pressing the Transfer button once.
Ip domain-name ipexpert.com|network-clock-participate wic 0|isdn switch-type primary-ni||controller t1 0/0/0|pri-group timeslots 1-3 - 24 service mgcp|no shut||int serial 0/0/0:23|isdn bind-l3 ccm-manager|ccm-manager mgcp|ccm-manager redundant-host 1
#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

39. When can you not use a Standard Local Route Group?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
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
When configuring TEHO.
8kb/s

40. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Precede the # with a > ... so 9011*>#
SERVICE PARAMETER> Automated Alternate Routing Enable > True
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True

41. Name 4 useful show commands for active calls.
24kb/s
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

42. How do you create a trunk on the switch side?
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

43. SNR is also known as?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Mobile Connect
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au

44. How can you confirm the MGCP GW is registered to CUCM - in IOS?
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
Mobile Voice Access
Sh ccm-manager
1 ... this is not optional!

45. How do you setup AutoRegistration in CUCM?
Mobile Connect
Dtmf-relay h245-alpha
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

46. What is CSIM?
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.
Call Simulator. You can use to validate path from router to the PSTN.
When configuring TEHO.
Service Parameter --> Block OffNet To OffNet Transfer --> True

47. How do you support multiple codecs on a dialpeer?
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
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
Apply voice-class codec
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

48. How do you allow H323 calls to be preserved should the primary H323 GW fail?
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.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

49. How do you change modes in voice register global?
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
Phone number followed by domain name.|i.e. 3006@ipxcme.com
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST

50. 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.
At the CLI: utils ntp status
SIP Dial Rules
To enable two-stage dialing.