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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. What terminology translates to AAR?
1 ... this is not optional!
Reroute when here is WAN congestion.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

2. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Call-manager-fallback or telephony-service mode SRST

3. Privacy is enabled system-wide in CUCM by default. (T or F)
Assign SIP Dial Rules
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Sh ip rsvp reservation||sh sccp connections
TRUE

4. Name 2 commands to verify RSVP functionality.
Call Simulator. You can use to validate path from router to the PSTN.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Sh ip rsvp reservation||sh sccp connections
Allows you to transfer by only pressing the Transfer button once.

5. How do you configure CUBE?

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6. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Need to Know||Anytime you have an external call - it needs to match on a route 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
SIP Route Pattern over a SIP Trunk.

7. What is the Enable Enterprise Feature Access Service Parameter used for?
To enable two-stage dialing.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
voice register pool|and|voice register dn
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.

8. How do you allow H323 calls to be preserved should the primary H323 GW fail?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
IP Voice Media Streaming App
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Re-enter the mgcp bind control and mgcp bind media commands.

9. What is the full E164 format?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
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

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

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11. How do you configure phone ports on an ESW?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
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

voice translation-profile|and|voice translation-rule

12. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Configure it on the route group through the route list - then it will be local to the route list.
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.
Re-enter the mgcp bind control and mgcp bind media commands.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

13. 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
Matches any length dialed number and truncates it to 4 digits.
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.
...
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

14. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
sh ephone reg|and|debug voip dialpeer
IP Voice Media Streaming App
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

15. What CUCM services should you activate?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Outbound dial-peers

16. 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
Sh ccm-manager
Debug ephone moh
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 create a trunk on the router side?
Auto Call Pickup Enabled
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
Sh ip rsvp reservation||sh sccp connections
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.

18. What does Display-IE do?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
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
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
Sends the Calling Name.

19. 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
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Service Parameters --> Mobile Voice Access Number
Sh ip rsvp reservation||sh sccp connections

20. How do you change modes in voice register global?
24kb/s
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
When they are explicitly matched in a destination-pattern in a dial-peer.
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST

21. On the RDP What is the Rerouting CSS used for?
Single Number Reach (Mobile Connect)
Dial-peers
To enable two-stage dialing.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

22. How is NTP sych setup 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.
CUCM OS Administration Settings --> NTP Servers
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Ccm-manager music-on-hold

23. How do you place SCCP and SIP phones into a single huntgroup?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Via the voice hunt-group parallel command
Apply voice-class codec
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

24. What are the basic SCCP Commands fro telephony-service in CUCME?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Single Number Reach (Mobile Connect)
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

25. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
sh ephone reg|and|debug voip dialpeer
voice register pool|and|voice register dn
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
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

26. How do you prioritize route groups?
Put them in a route list.
Single Number Reach (Mobile Connect)
Debug isdn q931
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

27. What are the basic ephone-dn and ephone commands?
(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
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
...
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

28. How do you configure DHCP in IOS?
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)
'Service Parameters --> Auto Answer Timer
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
(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

29. 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
#after-hours block pattern
Dtmf-relay h245-alpha
CUCM Service Parameters --> Call Park Reversion Timer

30. What would force you to use telephony-service to configure SRST?
#voice service voip| #no supplementary-service h225-notify cid-update
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Service Parameter --> Block OffNet To OffNet Transfer --> True
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

31. How do you ensure that G.711 only is used?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
#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

32. What are the 3 mandatory commands within call-manager-fallback?
Assign SIP Dial Rules
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 ....
'Use before the ... so XXXX
Max-dn|max-ephone|ip source-address

33. What are the commands to configure a MGCP Gateway? (router)
Dial-peers
Sends the Calling Name.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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

34. In CME Where is the Calling Name derived from?
In CUCM - configure CFUR to point to it's E164 number.
...
To enable two-stage dialing.
The name configuration field in ephone-dn and voice register dn

35. How do you verify that NTP is working on the CUCM server?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
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

36. 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
Auto Call Pickup Enabled
Assign SIP Dial Rules
Device Pool Locations

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

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

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39. 'If you want to make changes to any softkeys where do you do it?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
#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

40. Where can you assign the AAR Group?
Dtmf-relay h245-alpha
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
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

41. How do you disable KPML?
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
Device Pool Locations
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Assign SIP Dial Rules

42. 'If you're not getting a DHCP address from CUCM what then?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Auto Call Pickup Enabled
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable

43. How do you enable AAR system wide?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
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
(config)#voice register dialplan
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.

44. Name 4 useful show commands for active calls.
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Service Parameters --> Mobile Voice Access Number

45. How do you see the details of calls coming in and out of the PRI?
8kb/s
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Debug isdn q931
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

46. Is the order of the MRGs in an MRGL significant?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
(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
24kb/s

47. What commands are needed to configure the voice register pool in CME?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
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

CUCM Service Parameters --> Automated Alternate Routing Enable --> True

48. What are the steps to integrate CUPS with CUCM?
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
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 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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

49. Two useful troubleshooting commands for CUCME?
...
sh ephone reg|and|debug voip dialpeer
#after-hours block pattern
Sends the Calling Name.

50. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
To notify SIP Phones of NTP
The Called/Calling Transformations are superceded.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
When multiple sites use the same route pattern - do your digit manipulation on a route list.