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CCIE Voice Test

Instructions:

Answer 50 questions in 15 minutes.
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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. Name 4 useful show commands for active calls.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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
Show ccm-manager

2. What are the commands to manually configure an MGCP gateway?
CUCM OS Administration Settings --> NTP Servers
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.
Sends the Calling Name.
#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

3. Which takes precedence Device Locations or Device Pool Locations?
...
Max-dn|max-ephone|ip source-address
Dtmf-relay h245-alpha
Device Pool Locations

4. What terminology translates to AAR?
SIP Route Pattern over a SIP Trunk.
Reroute when here is WAN congestion.
On CUCM it's identical to adding an H323 GW.
Via the voice hunt-group parallel command

5. Is the order of the MRGs in an MRGL significant?
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.
Single Number Reach (Mobile Connect)
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

6. In CME Where is the Calling Name derived from?
The name configuration field in ephone-dn and voice register dn
#test voice translation rule 1 <input to test>
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
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.

7. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Max-dn|max-ephone|ip source-address
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

8. How do you configure CUCM redundancy on an H323 gateway?
(config)#sh cdp neigh detail
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
#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
No POTS dial-peer or no direct-inward-dial on a pots dial-peer

9. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Bottom up - to prevent glare.

#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

10. An MGCP gateway serving as an SRST router requires what?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Automatically configures the MGCP GW for you.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

11. What do you need to do to activate the CME GUI?
Auto Call Pickup Enabled
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
Top Down means that channel 1 will be the first channel used to place outgoing calls.
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

12. When using drop-through-option What is the max number of huntgroups?
TRUE
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
1 ... this is not optional!
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

13. '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.
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
1 ... this is not optional!
Re-enter the mgcp bind control and mgcp bind media commands.

14. When a device is roaming to a different site How does it get the Roaming Sensitive Settings from the Device Pool?
Assign a logout-profile to the ephone.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Show 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.

15. How do you prevent H323 caller-id updates to CUCM
Debug isdn q931
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
#voice service voip| #no supplementary-service h225-notify cid-update
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

16. What is the full E164 format?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
'Use before the ... so XXXX
At the CLI: utils ntp status
Matches any length dialed number and truncates it to 4 digits.

17. How much bandwidth does a G.729 call including layer 3 require?
IP Voice Media Streaming App
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 multiple sites use the same route pattern - do your digit manipulation on a route list.

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

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19. 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)
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
When they are explicitly matched in a destination-pattern in a dial-peer.

20. Describe how you configure SIP URI functionality.
(config)#sh cdp neigh detail

(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Ccm-manager music-on-hold

21. 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?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
The name configuration field in ephone-dn and voice register dn

22. How do you setup AutoRegistration in CUCM?
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.
Apply it to the ephone.
voice translation-profile|and|voice translation-rule
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

23. How do you disable KPML?
#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
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
Mobile Connect
Assign SIP Dial Rules

24. What are the commands to configure an H323 GW?
Sends the Calling Name.
#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
#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

25. How do you configure class of service (CoS) in CUCM? CME?
Automatically configures the MGCP GW for you.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Put them in a route list.

26. When setting up SIP URI where do you configure the CUCM's domain name?'
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
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
To enable two-stage dialing.
Mobile Connect

27. On the RDP What is the CSS used for?
Mobile Voice Access
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
#after-hours block pattern
Debug ephone moh

28. How do you configure phone ports on an ESW?
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
The Called/Calling Transformations are superceded.
Re-enter the mgcp bind control and mgcp bind media commands.
IP Voice Media Streaming App

29. What CM Service needs to be start in Serviceability for MOH to work?
Assign a logout-profile to the ephone.
IP Voice Media Streaming App
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
SIP Route Pattern over a SIP Trunk.

30. What is the Enable Enterprise Feature Access Service Parameter used for?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
To enable two-stage dialing.
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

31. How do you configure SRST?
Call-manager-fallback or telephony-service mode SRST
Dtmf-relay h245-alpha
Top Down means that channel 1 will be the first channel used to place outgoing calls.
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

32. '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> Automated Alternate Routing Enable > True
...
Reroute when here is WAN congestion.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

33. What does rule 1 /^.*(....$)/ /1/ do?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
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
Matches any length dialed number and truncates it to 4 digits.
Automatically configures the MGCP GW for you.

34. How do you see multicast packets being sent?
Debug ephone moh
Sends the Calling Name.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Matches any length dialed number and truncates it to 4 digits.

35. Where is it best to manipulate digits for inbound calls?

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36. Where can you assign the AAR Group?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
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
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

37. Name 2 commands to verify RSVP functionality.
(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
Debug isdn q931
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
Sh ip rsvp reservation||sh sccp connections

38. What does Display-IE do?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Sends the Calling Name.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

39. Describe the relationship between route patterns and end devices.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
(config)#voice register dialplan
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.

40. How do you use an ephone template?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Precede the # with a > ... so 9011*>#
voice translation-profile|and|voice translation-rule
Apply it to the ephone.

41. What would force you to use telephony-service to configure SRST?
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
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.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

42. What are the commands to set up the PRI?

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43. Calls coming from CUCM to PSTN need what?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Ccm-manager music-on-hold
Service Parameter --> Block OffNet To OffNet Transfer --> True
Outbound dial-peers

44. How do you use the # as a string terminator within a SIP Dial Rule?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Precede the # with a > ... so 9011*>#
1 ... this is not optional!
Sends the Calling Name.

45. How do you allow the Calling Name to be sent to the PSTN on a router?
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.
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
To notify SIP Phones of NTP
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie

46. What's the difference between AAR and SRST?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Call Simulator. You can use to validate path from router to the PSTN.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

47. 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
Phone number followed by domain name.|i.e. 3006@ipxcme.com
sh ephone reg|and|debug voip dialpeer
Sh ccm-manager

48. How do you block calls under call-manager-fallback?
TRUE
#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
#after-hours block pattern
Call forwarding between voip to voip (when CUBE is in play)

49. How do you support multiple codecs on a dialpeer?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
#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
Apply voice-class codec

50. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Outbound dial-peers
In the CUCM 7.0.1 Release Notes ... search for restore.