Test your basic knowledge |

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 much bandwidth does a G.729 call including layer 3 require?
#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
voice translation-profile|and|voice translation-rule
Matches any length dialed number and truncates it to 4 digits.
24kb/s

2. How do you prevent H323 caller-id updates to CUCM
...
#voice service voip| #no supplementary-service h225-notify cid-update
voice register pool|and|voice register dn
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files

3. In CME Where is the Calling Name derived from?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
The name configuration field in ephone-dn and voice register dn
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
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.

4. What are the commands to configure a SIP phone in CUCME?
#voice class h323 1| #h225 timeout tcp establish 3||#dial-peer voice 3000 voip| #destination-pattern 3...| # voice-class h323 1| #session target ipv4:10.1.5.3 (sub)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #pre

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
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

5. Calls coming from CUCM to PSTN need what?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Outbound dial-peers
IP Voice Media Streaming App

6. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
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.
Ccm-manager music-on-hold
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
(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

7. 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
Re-enter the mgcp bind control and mgcp bind media commands.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
When configuring TEHO.
...

8. SNR is also known as?
Mobile Connect
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Call Simulator. You can use to validate path from router to the PSTN.
(config)#sh cdp neigh detail

9. With the gw-priority command - does higher or lower priority take precedence?
Higher
Sh ccm-manager
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

10. '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

Sends the Calling Name.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
...

11. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
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
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
In the CUCM 7.0.1 Release Notes ... search for restore.

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
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Re-enter the mgcp bind control and mgcp bind media commands.
Via the voice hunt-group parallel command

13. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
SIP Route Pattern over a SIP Trunk.

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.

14. How can you confirm the MGCP GW is registered to CUCM - in IOS?
(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
#test voice translation rule 1 <input to test>
Sh ccm-manager
Debug ephone moh

15. How do you configure phone ports on a 3750?
Call Simulator. You can use to validate path from router to the PSTN.
#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
Sends the Calling Name.
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

16. What is SIP URI?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Dtmf-relay h245-alpha
(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
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable

17. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
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 ....

Bottom up - to prevent glare.

18. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
The Called/Calling Transformations are superceded.
#voice service voip| #no supplementary-service h225-notify cid-update
Sends the Calling Name.
SIP Route Pattern over a SIP Trunk.

19. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

20. How do you test a Voice Translation Rule?
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
Re-enter the mgcp bind control and mgcp bind media commands.
Configure it on the route group through the route list - then it will be local to the route list.
#test voice translation rule 1 <input to test>

21. How do you configure class of service (CoS) in CUCM? CME?
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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
#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.

22. How do you place SCCP and SIP phones into a single huntgroup?
Via the voice hunt-group parallel command
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
#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
It's best to strip digits at the voice port.

23. What does an H323 GW require that MGCP GWs do not?
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
Dial-peers
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.
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST

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

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25. When using drop-through-option What is the max number of huntgroups?
#after-hours block pattern
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
(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
1 ... this is not optional!

26. 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
Apply voice-class codec
To enable two-stage dialing.

27. What is call-forward pattern used for?
'Use before the ... so XXXX
Call forwarding between voip to voip (when CUBE is in play)
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
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

28. What does KPML do?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Show ccm-manager
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

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

30. How do you configure SRST?
#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
Call-manager-fallback or telephony-service mode SRST

Single Number Reach (Mobile Connect)

31. What is the Service Parameter to allow the phone that is picking ip to auto answer?
The Called/Calling Transformations are superceded.
Auto Call Pickup Enabled

Higher

32. What CM Service needs to be start in Serviceability for MOH to work?
Sends the Calling Name.
IP Voice Media Streaming App
Show ccm-manager
TRUE

33. What are two caveats to using the ccm-manager config server</ip> command?
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
Configure it on the route group through the route list - then it will be local to the route list.
Call Simulator. You can use to validate path from router to the PSTN.

34. What terminology translates to SRST?
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
Reroute when there is a WAN Outage.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
...

35. How do you prevent toll fraud on CME?
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 ....
Bandwidth
'Use before the ... so XXXX
Bottom up - to prevent glare.

36. When are digits stripped in a gateway?
When they are explicitly matched in a destination-pattern in a dial-peer.
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.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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

37. How do you prevent toll fraud on CUCM?
In CUCM - configure CFUR to point to it's E164 number.
SERVICE PARAMETER> Automated Alternate Routing Enable > True

SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True

38. What do you need to do to activate the CME GUI?
#after-hours block pattern
...
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
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

39. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Call-manager-fallback or telephony-service mode SRST
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.

40. When do you do digit manipulation at the route pattern as opposed to the route list?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Precede the # with a > ... so 9011*>#
When multiple sites use the same route pattern - do your digit manipulation on a route list.
On CUCM it's identical to adding an H323 GW.

41. How do you see multicast packets being sent?
Debug ephone moh
...
sh ephone reg|and|debug voip dialpeer
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

42. How do you configure DHCP in IOS?
Ccm-manager music-on-hold
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec 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
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.

43. What is a best practice for digit manipulation - in regards to H323 GWs?
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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
#voice class h323 1| #h225 timeout tcp establish 3||#dial-peer voice 3000 voip| #destination-pattern 3...| # voice-class h323 1| #session target ipv4:10.1.5.3 (sub)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #pre

44. How do you configure a SIP Trunk? (router side)
At the CLI: utils ntp status
In the CUCM 7.0.1 Release Notes ... search for restore.
Assign SIP Dial Rules
(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

45. On the RDP What is the CSS used for?
Mobile Voice Access
Call Simulator. You can use to validate path from router to the PSTN.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
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.

46. On the RDP What is the Rerouting CSS used for?
(config)#sh cdp neigh detail
Single Number Reach (Mobile Connect)
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Mobile Connect

47. What is the full E164 format?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
(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
Automatically configures the MGCP GW for you.
voice translation-profile|and|voice translation-rule

48. Where do you use VIA zone?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
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

49. 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.
In CUCM - configure CFUR to point to it's E164 number.
Assign a logout-profile to the ephone.
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

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

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