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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 verify that NTP is working on the CUCM server?
Assign a logout-profile to the ephone.
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
At the CLI: utils ntp status
To enable two-stage dialing.

2. How do you allow the Calling Name to be sent to the PSTN on a router?
Configure it on the route group through the route list - then it will be local to the route list.
(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
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)

3. What is the bit rate for a G.729 call excluding layer 2?
8kb/s
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Assign SIP Dial Rules

4. Name 4 useful show commands for active calls.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
(config)#voice register dialplan
Master/Slave relationship. CUCM controls it.

5. How do you create a trunk on the router side?
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
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
TRUE
(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

6. How do you create a trunk on the switch side?
Call forwarding between voip to voip (when CUBE is in play)
Sh ip rsvp reservation||sh sccp connections
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
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

7. What CUCM services should you activate?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Precede the # with a > ... so 9011*>#

8. 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
TRUE

(config)#sh cdp neigh detail

9. What commands are needed to configure the voice register pool in CME?
...
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
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

10. How do you setup AutoRegistration in CUCM?
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
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Dtmf-relay h245-alpha
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

11. How do you configure an MGCP GW? (router side)
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
(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
#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
Debug ephone moh

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

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13. When do you use translate called? Translate calling?
Configure it on the route group through the route list - then it will be local to the route list.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.

14. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?
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
It's best to strip digits at the voice port.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
To enable two-stage dialing.

15. What is Mobile Voice Access?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk 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
#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
(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

16. 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
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
SIP Route Pattern over a SIP Trunk.
Call-manager-fallback or telephony-service mode SRST

17. What is a best practice for digit manipulation - in regards to H323 GWs?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

18. What's the best way to perform digit manipulation on a route group?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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
Dial-peers
Configure it on the route group through the route list - then it will be local to the route list.

19. How is NTP sych setup in CUCM?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
CUCM OS Administration Settings --> NTP Servers
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

20. What are the basic ephone-dn and ephone commands?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
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

21. What should you configure before entering auto qos?
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
...
IP Voice Media Streaming App
Bandwidth

22. Where do you use VIA zone?
CUCM Service Parameters --> Call Park Reversion Timer
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
Higher
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

23. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
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 .)
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Ccm-manager music-on-hold
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

24. Where can you assign the AAR Group?
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.
...
(config)#sh cdp neigh detail
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

25. How do you set up redundancy on outbound dial-peers on an H323 gateway?
Device Pool Locations
24kb/s
...
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

26. How do you support multiple codecs on a dialpeer?
#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
Apply voice-class codec
At the CLI: utils ntp status
Allows you to transfer by only pressing the Transfer button once.

27. 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?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
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
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.

28. How do you use an ephone template?

Apply it to the ephone.
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
Reroute when there is a WAN Outage.

29. In Gatekeeper CAC how do you restrict a specific endpoint?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
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

30. What are the commands to configure a T1/E1 PRI?
(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
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
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 ....
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. How do you set the Call Park Reversion Timer?
Apply voice-class codec
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
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)
CUCM Service Parameters --> Call Park Reversion Timer

32. What are the basic SCCP Commands fro telephony-service in CUCME?
Mobile Connect
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.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Service Parameter --> Block OffNet To OffNet Transfer --> True

33. How do you prevent H323 caller-id updates to CUCM
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
#voice service voip| #no supplementary-service h225-notify cid-update
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

34. What are the commands to configure a SIP phone in CUCME?
#after-hours block pattern
#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
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

35. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
...
Sends the Calling Name.
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

36. How do you change modes in voice register global?
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
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
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
(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

37. How do you configure CUCM redundancy on an H323 gateway?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
#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
#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
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.

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

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39. When are digits stripped in a gateway?
...
Debug isdn q931
When they are explicitly matched in a destination-pattern in a dial-peer.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

40. How do you set the inter-digit timeout for SIP phones in CME?
(config)#voice register dialplan
Outbound dial-peers
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

41. What are the base telephony-service commands for CME?

Automatically configures the MGCP GW for you.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

42. What are the commands to create vlans on an ESW?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

43. How do you configure a SIP Trunk? (router side)
Bottom up - to prevent glare.
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
(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

44. What dtmf-relay type do you use for an H323 GW?
voice translation-profile|and|voice translation-rule
Outbound dial-peers
Dtmf-relay h245-alpha
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

45. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Bottom up - to prevent glare.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Sh ip rsvp reservation||sh sccp connections
CUCM Service Parameters --> Automated Alternate Routing Enable --> True

46. When a device is roaming to a different site How does it get the Roaming Sensitive Settings from the Device Pool?
Sends the Calling Name.
'Use before the ... so XXXX
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer

47. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
#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
Reroute when here is WAN congestion.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
...

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.
When multiple sites use the same route pattern - do your digit manipulation on a route list.

49. How much bandwidth does a G.729 call including layer 3 require?
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)
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
sh ephone reg|and|debug voip dialpeer
24kb/s

50. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Sh ccm-manager
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.
To enable two-stage dialing.