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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. How do you set the inter-digit timeout for SIP phones in CME?
Assign SIP Dial Rules
At the CLI: utils ntp status
(config)#voice register dialplan
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

2. What are the steps to configure Single Number Reach and Mobile Voice Access?
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

1 ... this is not optional!
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

3. What is the Enable Enterprise Feature Access Service Parameter used for?
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
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
...
To enable two-stage dialing.

4. What kind of relationship does an MGCP gateway have with CUCM?
Mobile Connect
Master/Slave relationship. CUCM controls it.
When they are explicitly matched in a destination-pattern in a dial-peer.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

5. In Gatekeeper CAC how do you restrict a specific endpoint?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Sends the Calling Name.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
(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

6. How do you see multicast packets being sent?
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
Debug ephone moh
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.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!

7. What should you configure before entering auto qos?
Bandwidth
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
Assign SIP Dial Rules
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

8. What would force you to use telephony-service to configure SRST?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
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.

9. 'If you want to make changes to any softkeys where do you do it?
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
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.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.

10. How do you support multiple codecs on a dialpeer?
It's best to strip digits at the voice port.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Apply voice-class codec
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.

11. What types of digit manipulation can you perform at the route pattern?
Debug isdn q931
Bottom up - to prevent glare.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

12. How do you use the # as a string terminator within a SIP Dial Rule?
Precede the # with a > ... so 9011*>#
Via the voice hunt-group parallel command
Service Parameters --> Mobile Voice Access Number
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True

13. How do you configure a gateway to register with gatekeeper?
Debug ephone moh
#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
Bottom up - to prevent glare.
To enable two-stage dialing.

14. What are two caveats to using the ccm-manager config server</ip> command?

(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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
The Called/Calling Transformations are superceded.

15. How do you configure class of service (CoS) in CUCM? CME?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Reroute when here is WAN congestion.

16. Is the + character supported on a VOIP dial-peer?

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17. How do you prevent toll fraud on CUCM?
Single Number Reach (Mobile Connect)
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
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.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True

18. 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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
...

19. When a device is roaming to a different site How does it get the Roaming Sensitive Settings from the Device Pool?
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
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
When they are explicitly matched in a destination-pattern in a dial-peer.

20. How much bandwidth does a G.729 call including layer 3 require?
24kb/s
Call Simulator. You can use to validate path from router to the PSTN.
At the CLI: utils ntp status
Show ccm-manager

21. 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
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
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
...

22. What are the commands to configure a MGCP Gateway? (router)
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
voice register pool|and|voice register dn
Debug ephone moh
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

23. 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
24kb/s
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

24. What are the commands to configure a T1/E1 PRI?
IP Voice Media Streaming App
CUCM Service Parameters --> Call Park Reversion Timer
When configuring TEHO.
(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

25. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
SIP Route Pattern over a SIP Trunk.
Service Parameters --> Mobile Voice Access Number
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 ipv4:10.1.5.3 (sub)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #pre

26. What's the difference between AAR and SRST?
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
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
1 ... this is not optional!

27. How do you enable AAR?
Assign a logout-profile to the ephone.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
SERVICE PARAMETER> Automated Alternate Routing Enable > True
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

28. Two useful troubleshooting commands for CUCME?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
#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 translation-profile|and|voice translation-rule
sh ephone reg|and|debug voip dialpeer

29. What are the steps to integrate CUPS with CUCM?
#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
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
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

30. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
(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
Re-enter the mgcp bind control and mgcp bind media commands.
Higher

31. Which takes precedence Device Locations or Device Pool Locations?
In CUCM - configure CFUR to point to it's E164 number.
Sh ip rsvp reservation||sh sccp connections
Device Pool Locations
8kb/s

32. 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?
(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
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
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
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.

33. What dtmf-relay type do you use for an H323 GW?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Dtmf-relay h245-alpha

34. What are the commands to configure SRST in fallback?
#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
#test voice translation rule 1 <input to test>
To enable two-stage dialing.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

35. What's the best way to do digit manipulation on an IOS gateway?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
voice translation-profile|and|voice translation-rule
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
'Service Parameters --> Auto Answer Timer

36. What's the best way to perform digit manipulation on a route group?
Configure it on the route group through the route list - then it will be local to the route list.
When they are explicitly matched in a destination-pattern in a dial-peer.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
#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

37. An MGCP gateway serving as an SRST router requires what?
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
(config)#voice register dialplan

38. When do you use translate called? Translate calling?
Precede the # with a > ... so 9011*>#
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

39. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
(config)#sh cdp neigh detail
#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
(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
Bottom up - to prevent glare.

40. Where do you use VIA zone?
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
#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
Master/Slave relationship. CUCM controls it.
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST

41. What commands are needed to configure the voice register pool in CME?
Dial-peers
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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
#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

42. What does the Transfer On-hook Enabled Service Parameter do?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
When configuring TEHO.
Allows you to transfer by only pressing the Transfer button once.

43. Name 4 useful show commands for active calls.
Precede the # with a > ... so 9011*>#
When they are explicitly matched in a destination-pattern in a dial-peer.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Bottom up - to prevent glare.

44. What does ccm-manager config server [IP]</ip> do?
Automatically configures the MGCP GW for you.
When they are explicitly matched in a destination-pattern in a dial-peer.
Put them in a route list.
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

45. What are the commands to configure an H323 GW?
(config)#voice register dialplan
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
#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

46. When a CUCM device dials a number - what happens?
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 .)
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Assign SIP Dial Rules
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

47. On the RDP What is the CSS used for?
Automatically configures the MGCP GW for you.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Mobile Voice Access
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

48. Does CUCM support RSVP natively?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Apply it to the ephone.
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
#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

49. When using drop-through-option What is the max number of huntgroups?
Single Number Reach (Mobile Connect)
Mobile Connect
When configuring TEHO.
1 ... this is not optional!

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

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CCIE Voice Test

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