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. What is Phone NTP Reference used for?
SIP Dial Rules
When configuring TEHO.
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

2. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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.
voice translation-profile|and|voice translation-rule

3. How do do configure TEHO?
Reroute when there is a WAN Outage.
TRUE
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
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

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

5. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
In the CUCM 7.0.1 Release Notes ... search for restore.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Apply it to the ephone.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True

6. What does ccm-manager config server [IP]</ip> do?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Automatically configures the MGCP GW for you.
Precede the # with a > ... so 9011*>#
SERVICE PARAMETER> Automated Alternate Routing Enable > True

7. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
voice register pool|and|voice register dn
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable

8. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
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
24kb/s

9. What is a best practice for digit manipulation - in regards to H323 GWs?
Apply it to the ephone.
(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 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.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

10. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.

11. What is the Service Parameter to allow the phone that is picking ip to auto answer?
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.
Auto Call Pickup Enabled
#test voice translation rule 1 <input to test>
#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

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

13. What is the Enable Enterprise Feature Access Service Parameter used for?
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
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
Matches any length dialed number and truncates it to 4 digits.
To enable two-stage dialing.

14. Does CUCM support RSVP natively?
Automatically configures the MGCP GW for you.
Dial-peers
sh ephone reg|and|debug voip dialpeer
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.

15. What are the commands to configure a SIP phone in CUCME?
(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
#test voice translation rule 1 <input to test>
Default is SRST||mode cme changes to CME|no mode cme changes back 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

16. How do you configure phone ports on a 3750?
The Called/Calling Transformations are superceded.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
It's best to strip digits at the voice port.

17. 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
Re-enter the mgcp bind control and mgcp bind media commands.
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

18. How do you configure CUBE?

19. Where do you use VIA zone?
Call-manager-fallback or telephony-service mode SRST
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
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

20. How do you verify that NTP is working on the CUCM server?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
At the CLI: utils ntp status
(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
#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

21. How do you verify where MOH is being served up from?

22. How do you disable KPML?
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Assign SIP Dial Rules
(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
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

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

24. What CUCM services should you activate?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
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
voice register pool|and|voice register dn
Mobile Voice Access

25. SNR is also known as?
Mobile Connect
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

26. When do you use translate called? Translate calling?
Call Simulator. You can use to validate path from router to the PSTN.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
...

27. What kind of relationship does an MGCP gateway have with CUCM?
Master/Slave relationship. CUCM controls it.
The Called/Calling Transformations are superceded.
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 ....
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

28. What's the best way to do digit manipulation on an IOS gateway?
(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
voice translation-profile|and|voice translation-rule
In the CUCM 7.0.1 Release Notes ... search for restore.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

29. How can you confirm the MGCP GW is registered to CUCM - in IOS?
#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
Sh ccm-manager
voice translation-profile|and|voice translation-rule
#voice service voip| #no supplementary-service h225-notify cid-update

30. What does Display-IE do?
8kb/s
Sends the Calling Name.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Max-dn|max-ephone|ip source-address

31. How do you block an external call from being transferred back out to the pstn by an internal user?
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Service Parameter --> Block OffNet To OffNet Transfer --> True
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

32. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

33. What are the commands to create the L3 routing interface for VLANS (SVI)?
24kb/s
(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
(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 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

34. If you're using Called/Calling Party Transformations at the Route Pattern and tranformations at the route list ... What does the RP affect and What does the RL affect?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
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
...

35. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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.
(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.

36. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
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
The name configuration field in ephone-dn and voice register dn
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

37. How do you enable AAR system wide?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
(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
Higher

38. What's the difference between AAR and SRST?
The Called/Calling Transformations are superceded.
Dial-peers
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Assign a logout-profile to the ephone.

39. How do you prevent toll fraud on CME?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
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 ....
Max-dn|max-ephone|ip source-address
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

40. Is the order of the MRGs in an MRGL significant?
In CUCM - configure CFUR to point to it's E164 number.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
To enable two-stage dialing.

41. An MGCP gateway serving as an SRST router requires what?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
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 .)
Dtmf-relay h245-alpha
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

42. How do you configure SRST?
#test voice translation rule 1 <input to test>
Single Number Reach (Mobile Connect)
Outbound dial-peers
Call-manager-fallback or telephony-service mode SRST

43. 'How do you inform a SIP phone of NTP information?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
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
CUCM OS Administration Settings --> NTP Servers
Call Simulator. You can use to validate path from router to the PSTN.

44. With the gw-priority command - does higher or lower priority take precedence?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Higher
CUCM Service Parameters --> Automated Alternate Routing Enable --> True

45. What are the commands to configure SRST in fallback?
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
Higher
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Debug ephone moh

46. How do you configure a gateway to register with gatekeeper?
(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
voice register pool|and|voice register dn
Reroute when here is WAN congestion.
#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

47. What should you configure before entering auto qos?
Bandwidth
To notify SIP Phones of NTP
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 .)
Assign a logout-profile to the ephone.

48. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie

49. What are the commands to configure NTP in IOS?
#voice service voip| #no supplementary-service h225-notify cid-update
#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
24kb/s
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

50. On an H323 GW - how do you adjust the timers for redundancy hunting?
To enable two-stage dialing.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
(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
Allows you to transfer by only pressing the Transfer button once.