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CCIE Voice Test

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Answer 50 questions in 15 minutes.
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Match each statement with the correct term.
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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 setup AutoRegistration in CUCM?
Ccm-manager music-on-hold
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

2. What's the difference between AAR and 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
Service Parameters --> Mobile Voice Access Number
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
(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

3. How do you test a Voice Translation Rule?
#test voice translation rule 1 <input to test>
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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.

4. If you can't use a tech-prefix or default tech-prefix how do you get around the fact that CUCM does not register it's endpoint to GK?
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
The Called/Calling Transformations are superceded.
#voice service voip| #no supplementary-service h225-notify cid-update
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)

5. Aside from the Media Resources --> Mobile Voice Access where else do you need to specify the the Mobile Voice Access DN?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Service Parameters --> Mobile Voice Access Number
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Mobile Connect

6. When do you do digit manipulation at the route pattern as opposed to the route list?
Call-manager-fallback or telephony-service mode SRST
(config)#sh cdp neigh detail
When multiple sites use the same route pattern - do your digit manipulation on a route list.
voice register pool|and|voice register dn

7. How do you configure CUCM redundancy on an H323 gateway?
Reroute when there is a WAN Outage.
Outbound dial-peers

#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

8. On the RDP What is the CSS used for?
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.
Mobile Voice Access
(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

9. What terminology translates to SRST?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Call forwarding between voip to voip (when CUBE is in play)
Reroute when there is a WAN Outage.

10. What does the Transfer On-hook Enabled Service Parameter do?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
The Called/Calling Transformations are superceded.
Allows you to transfer by only pressing the Transfer button once.
Assign SIP Dial Rules

11. How can you confirm the MGCP GW is registered to CUCM - in IOS?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Sh ccm-manager

12. SNR is also known as?
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
On CUCM it's identical to adding an H323 GW.
Mobile Connect
Via the voice hunt-group parallel command

13. What do you need to do to activate the CME GUI?
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
When configuring TEHO.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer

14. How would you enable security on a GK?
The name configuration field in ephone-dn and voice register dn
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 register pool|and|voice register dn
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

15. How do you prevent H323 caller-id updates to CUCM
8kb/s
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
#voice service voip| #no supplementary-service h225-notify cid-update
Mobile Connect

16. What are the commands to create vlans on an ESW?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Apply it to the ephone.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

17. In CME Where is the Calling Name derived from?
Automatically configures the MGCP GW for you.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
The name configuration field in ephone-dn and voice register dn
Call Simulator. You can use to validate path from router to the PSTN.

18. How do you enable AAR?
Master/Slave relationship. CUCM controls it.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Reroute when there is a WAN Outage.
Single Number Reach (Mobile Connect)

19. What types of digit manipulation can you perform at the route pattern?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Ccm-manager music-on-hold

20. How do you configure a SIP Trunk? (router side)
(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.
Via the voice hunt-group parallel command
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

21. How do you verify that NTP is working on the CUCM server?
At the CLI: utils ntp status
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
...
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.

22. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
Auto Call Pickup Enabled
The name configuration field in ephone-dn and voice register dn
Debug ephone moh
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

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

Warning: Invalid argument supplied for foreach() in /var/www/html/basicversity.com/show_quiz.php on line 183

24. How do you configure AAR?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Bandwidth
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

25. What are the base telephony-service commands for CME?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
To enable two-stage dialing.
Call-manager-fallback or telephony-service mode SRST
Via the voice hunt-group parallel command

26. How much bandwidth does a G.729 call including layer 3 require?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
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)
#after-hours block pattern
24kb/s

27. What are the 3 mandatory commands within call-manager-fallback?

In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Max-dn|max-ephone|ip source-address

28. 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 os truly unsurpassed. I have opened up a wealth of opportunities for myself and my family and
...
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)#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)#sh cdp neigh detail

29. How do you use an ephone template?
Apply it to the ephone.
Outbound dial-peers
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. What are the commands to 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
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 ....
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
#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

31. What CM Service needs to be start in Serviceability for MOH to work?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
IP Voice Media Streaming App
CUCM Service Parameters --> Call Park Reversion Timer
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

32. How do you ensure that G.711 only is used?
The Called/Calling Transformations are superceded.
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 .)
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Device Pool Locations

33. How do you set the Call Park Reversion Timer?
Mobile Connect
Debug isdn q931
CUCM Service Parameters --> Call Park Reversion Timer
Debug ephone moh

34. Describe how you configure SIP URI functionality.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

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

35. What are the steps to integrate CUPS with CUCM?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
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
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

36. What are two caveats to using the ccm-manager config server</ip> command?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

#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
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

37. What commands are needed to configure the voice register pool in CME?
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)
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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Higher

38. How do you block calls under call-manager-fallback?
#after-hours block pattern
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Put them in a route list.
(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

39. How do you allow H323 calls to be preserved should the primary H323 GW fail?
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
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

40. What CUCM services should you activate?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.

Outbound dial-peers

41. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
voice register pool|and|voice register dn
The Called/Calling Transformations are superceded.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Via the voice hunt-group parallel command

42. 'If you're not getting a DHCP address from CUCM what then?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
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)
Auto Call Pickup Enabled
Call-manager-fallback or telephony-service mode SRST

43. Which takes precedence Device Locations or Device Pool Locations?
Device Pool Locations
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.
Show ccm-manager
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.

44. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
SIP Route Pattern over a SIP Trunk.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
#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

45. How do you enable Extension Mobility for a device in CME?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
The Called/Calling Transformations are superceded.
TRUE
Assign a logout-profile to the ephone.

46. What are the steps to configure Single Number Reach and Mobile Voice Access?
Higher
voice translation-profile|and|voice translation-rule
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.

47. What should you configure before entering auto qos?
8kb/s
SIP Dial Rules
Bandwidth
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

48. When a CUCM device dials a number - what happens?
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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
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 .)
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

49. What is the Enable Enterprise Feature Access Service Parameter used for?
To enable two-stage dialing.
Service Parameter --> Block OffNet To OffNet Transfer --> True
SIP Route Pattern over a SIP Trunk.
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.

50. Two useful troubleshooting commands for CUCME?
#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
sh ephone reg|and|debug voip dialpeer
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

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CCIE Voice Test

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