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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 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.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True

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

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3. On the RDP What is the Rerouting CSS used for?
Single Number Reach (Mobile Connect)
voice register pool|and|voice register dn
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au

4. 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
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Single Number Reach (Mobile Connect)

5. 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?
Outbound dial-peers
(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
When multiple sites use the same route pattern - do your digit manipulation on a route list.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.

6. What's the best way to do digit manipulation on an IOS gateway?
voice translation-profile|and|voice translation-rule
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
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

7. On the RDP What is the CSS used for?
Bottom up - to prevent glare.
Mobile Voice Access
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Service Parameter --> Block OffNet To OffNet Transfer --> True

8. What does ccm-manager config server [IP]</ip> do?
Automatically configures the MGCP GW for you.
#after-hours block pattern
Dtmf-relay h245-alpha
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

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

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10. What is Phone NTP Reference used for?
Re-enter the mgcp bind control and mgcp bind media commands.
Service Parameter --> Block OffNet To OffNet Transfer --> True
On CUCM it's identical to adding an H323 GW.
To notify SIP Phones of NTP

11. What are the base telephony-service commands for CME?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Higher
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

12. What does rule 1 /^.*(....$)/ /1/ do?

Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Allows you to transfer by only pressing the Transfer button once.
Matches any length dialed number and truncates it to 4 digits.

13. When setting up SIP URI where do you configure the CUCM's domain name?'
24kb/s
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
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.
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name

14. What's the difference between AAR and SRST?
SIP Dial Rules
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Reroute when here is WAN congestion.
#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

15. How would you enable security on a GK?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
1 ... this is not optional!
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

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

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17. What does KPML do?
When configuring TEHO.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
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.
Service Parameter --> Block OffNet To OffNet Transfer --> True

18. What CM Service needs to be start in Serviceability for MOH to work?
At the CLI: utils ntp status
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
IP Voice Media Streaming App
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.

19. Does CUCM support RSVP natively?
#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
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.
Assign SIP Dial Rules
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.

20. How do you setup AutoRegistration in CUCM?
Higher
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable

21. Maximum Wait Time for Desk Pickup?
1 ... this is not optional!
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
8kb/s

22. On an MGCP GW - how could you see the primary and backup CUCM servers?
(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
Mobile Voice Access
Call forwarding between voip to voip (when CUBE is in play)
Show ccm-manager

23. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Ccm-manager music-on-hold
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
(config)#sh cdp neigh detail
Reroute when there is a WAN Outage.

24. How do you get around relying on DNS for your CUCMs?
24kb/s
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Reroute when here is WAN congestion.

25. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Reroute when there is a WAN Outage.
SIP Dial Rules
Ccm-manager music-on-hold

26. How do you enable AAR system wide?

Outbound dial-peers
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

27. How do you allow the Calling Name to be sent to the PSTN on a router?
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
TRUE
8kb/s
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

28. 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.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

29. Name 4 useful show commands for active calls.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
At the CLI: utils ntp status
Apply it to the ephone.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

30. When a device is roaming to a different site How does it get the Roaming Sensitive Settings from the Device Pool?
voice register pool|and|voice register dn
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Automatically configures the MGCP GW for you.
(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

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

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32. How do you allow H323 calls to be preserved should the primary H323 GW fail?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
(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 --> True

33. How do you create a trunk on the switch side?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
The Called/Calling Transformations are superceded.
Allows you to transfer by only pressing the Transfer button once.
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

34. What is CSIM?
Call Simulator. You can use to validate path from router to the PSTN.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Dtmf-relay h245-alpha
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

35. How do you prevent toll fraud on CUCM?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Single Number Reach (Mobile Connect)
Service Parameters --> Mobile Voice Access Number

36. How do you enable AAR?
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Re-enter the mgcp bind control and mgcp bind media commands.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
CUCM OS Administration Settings --> NTP Servers

37. How do you configure SRST?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Debug isdn q931
Call-manager-fallback or telephony-service mode SRST
Master/Slave relationship. CUCM controls it.

38. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
...
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
#voice service voip| #no supplementary-service h225-notify cid-update

39. 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.
Call forwarding between voip to voip (when CUBE is in play)
Top Down means that channel 1 will be the first channel used to place outgoing calls.

40. What is the full E164 format?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Configure it on the route group through the route list - then it will be local to the route list.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Mobile Connect

41. 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)
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Sh ip rsvp reservation||sh sccp connections
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

42. 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
Mobile Connect
...
#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

43. How do you configure AAR?
Call forwarding between voip to voip (when CUBE is in play)
Sh ccm-manager
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Assign SIP Dial Rules

44. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
The Called/Calling Transformations are superceded.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Max-dn|max-ephone|ip source-address

45. Privacy is enabled system-wide in CUCM by default. (T or F)
Dtmf-relay h245-alpha
Debug isdn q931
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
TRUE

46. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?
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
Service Parameter --> Block OffNet To OffNet Transfer --> True
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
To notify SIP Phones of NTP

47. 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
Matches any length dialed number and truncates it to 4 digits.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
At the CLI: utils ntp status

48. How do you block calls under call-manager-fallback?
When configuring TEHO.
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
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
#after-hours block pattern

49. What does the Transfer On-hook Enabled Service Parameter do?
Allows you to transfer by only pressing the Transfer button once.
The name configuration field in ephone-dn and voice register dn
When they are explicitly matched in a destination-pattern in a dial-peer.
SIP Route Pattern over a SIP Trunk.

50. How do you configure DHCP in IOS?
#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
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
(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
24kb/s