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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
Dial-peers
voice translation-profile|and|voice translation-rule
Single Number Reach (Mobile Connect)

2. 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
...

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
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

3. How do you enable Extension Mobility for a device in CME?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
#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
Assign a logout-profile to the ephone.

4. How do do configure TEHO?
Assign SIP Dial Rules
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
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
Mobile Connect

5. Maximum Wait Time for Desk Pickup?
Reroute when there is a WAN Outage.
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
In the CUCM 7.0.1 Release Notes ... search for restore.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.

6. Aside from the Media Resources --> Mobile Voice Access where else do you need to specify the the Mobile Voice Access DN?
Service Parameters --> Mobile Voice Access Number
...
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

7. An MGCP gateway serving as an SRST router requires what?
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
IP Voice Media Streaming App
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Redundant dial plan. One on CUCM and one on the routers via dial-peers.

8. How do you test a Voice Translation Rule?
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
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 .)
#test voice translation rule 1 <input to test>
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)

9. What are the commands to configure SIP phones in CME?
(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
(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
The Called/Calling Transformations are superceded.
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

10. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Re-enter the mgcp bind control and mgcp bind media commands.
Service Parameter --> Block OffNet To OffNet Transfer --> True
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

11. Describe the relationship between route patterns and end devices.
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.
Put them in a route list.
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.
8kb/s

12. How do you configure an MGCP GW? (router side)
(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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
#after-hours block pattern
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

13. 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?
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
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

14. 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?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
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.
#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
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.

15. How do you configure a SIP Trunk? (router side)
(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
Mobile Voice Access
(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
Sends the Calling Name.

16. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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
It's best to strip digits at the voice port.

Need to Know||Anytime you have an external call - it needs to match on a route pattern.

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

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18. What kind of relationship does an MGCP gateway have with CUCM?
To enable two-stage dialing.
Master/Slave relationship. CUCM controls it.
Debug ephone moh
SERVICE PARAMETER> Automated Alternate Routing Enable > True

19. What is the bit rate for a G.729 call excluding layer 2?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
8kb/s

20. What terminology translates to SRST?
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.
Reroute when there is a WAN Outage.
#voice service voip| #no supplementary-service h225-notify cid-update
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

21. How do you configure phone ports on an ESW?
Dtmf-relay h245-alpha
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
Max-dn|max-ephone|ip source-address
SERVICE PARAMETER> Automated Alternate Routing Enable > True

22. What is SIP URI?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
The name configuration field in ephone-dn and voice register dn

23. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
Bottom up - to prevent glare.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Call Simulator. You can use to validate path from router to the PSTN.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

24. What are the commands to manually configure an MGCP gateway?
24kb/s
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-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
#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

25. How do you configure AAR?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Dial-peers
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.

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

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27. What are the commands to create the L3 routing interface for VLANS (SVI)?
Service Parameter --> Block OffNet To OffNet Transfer --> True
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
(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 Connect

28. How do you enable AAR system wide?
Outbound dial-peers
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Reroute when here is WAN congestion.
Auto Call Pickup Enabled

29. What is call-forward pattern used for?
To enable two-stage dialing.
Call forwarding between voip to voip (when CUBE is in play)

Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

30. Where can you assign the AAR Group?
Device Pool Locations
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
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

31. What is the full E164 format?
It's best to strip digits at the voice port.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
When configuring TEHO.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer

32. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
SIP Route Pattern over a SIP Trunk.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

33. How do you place SCCP and SIP phones into a single huntgroup?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Via the voice hunt-group parallel command
(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
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

34. What are the 3 mandatory commands within call-manager-fallback?
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
To enable two-stage dialing.
#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
Max-dn|max-ephone|ip source-address

35. How would you enable security on a GK?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
The name configuration field in ephone-dn and voice register dn
On CUCM it's identical to adding an H323 GW.
'Use before the ... so XXXX

36. How do you set the inter-digit timeout for SIP phones in CME?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
(config)#voice register dialplan
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
SIP Dial Rules

37. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
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 .)
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
24kb/s

38. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
TRUE
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Ccm-manager music-on-hold
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

39. When a device is roaming to a different site How does it get the Roaming Sensitive Settings from the Device Pool?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Debug ephone moh
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

40. What does Display-IE do?
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
Service Parameters --> Mobile Voice Access Number
The Called/Calling Transformations are superceded.
Sends the Calling Name.

41. How do you set up redundancy on outbound dial-peers on an H323 gateway?
The name configuration field in ephone-dn and voice register dn
Bandwidth
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

42. How do you get around relying on DNS for your CUCMs?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Debug ephone moh
Apply it to the ephone.

43. 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
TRUE
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
...
Call forwarding between voip to voip (when CUBE is in play)

44. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?

#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Mobile Voice Access

45. In CME Where is the Calling Name derived from?
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 PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
The name configuration field in ephone-dn and voice register dn
Show ccm-manager

46. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
Redundant dial plan. One on CUCM and one on the routers via 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.
It's best to strip digits at the voice port.

47. What is the Enable Enterprise Feature Access Service Parameter used for?
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
To enable two-stage dialing.
At the CLI: utils ntp status
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

48. What should you configure before entering auto qos?
The Called/Calling Transformations are superceded.
Higher
Call Simulator. You can use to validate path from router to the PSTN.
Bandwidth

49. How do you prioritize route groups?
(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
#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
Put them in a route list.
Service Parameters --> Mobile Voice Access Number

50. How do you verify that NTP is working on the CUCM server?
(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
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.
At the CLI: utils ntp status
Call-manager-fallback or telephony-service mode SRST