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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.
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 configure SRST?
IP Voice Media Streaming App
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Bottom up - to prevent glare.
Call-manager-fallback or telephony-service mode SRST

2. What does Display-IE do?
SIP Route Pattern over a SIP Trunk.
Device Pool Locations

Sends the Calling Name.

3. 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?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
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)
#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
Apply it to the ephone.

4. How do you allow H323 calls to be preserved should the primary H323 GW fail?
'Service Parameters --> Auto Answer Timer
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
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

5. Two useful troubleshooting commands for CUCME?
sh ephone reg|and|debug voip dialpeer
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
In the CUCM 7.0.1 Release Notes ... search for restore.
Master/Slave relationship. CUCM controls it.

6. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
SIP Route Pattern over a SIP Trunk.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Sh ip rsvp reservation||sh sccp connections

7. How do you disable KPML?
Assign SIP Dial Rules
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Redundant dial plan. One on CUCM and one on the routers via dial-peers.

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

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

9. 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
Auto Call Pickup Enabled
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
(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

10. How do you see the details of calls coming in and out of the PRI?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Debug isdn q931
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
TRUE

11. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Sh ccm-manager
Allows you to transfer by only pressing the Transfer button once.

12. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Sh ip rsvp reservation||sh sccp connections
#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
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

13. How do you verify that NTP is working on the CUCM server?
The name configuration field in ephone-dn and voice register dn
#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
At the CLI: utils ntp status
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

14. Which takes precedence Device Locations or Device Pool Locations?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Device Pool Locations
...
#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

15. What are the 3 mandatory commands within call-manager-fallback?
Reroute when there is a WAN Outage.
Max-dn|max-ephone|ip source-address
Call-manager-fallback or telephony-service mode SRST
Bottom up - to prevent glare.

16. How do you configure CUCM redundancy on an H323 gateway?
#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 every digit to CUCM in real-time with digit-analysis being handled by CUCM.
#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
It's best to strip digits at the voice port.

17. 'How do you inform a SIP phone of NTP information?
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 > 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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
'Use before the ... so XXXX

18. What are the commands to configure SRST in fallback?
8kb/s
Outbound dial-peers
Master/Slave relationship. CUCM controls it.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

19. What is a best practice for digit manipulation - in regards to H323 GWs?
Debug ephone moh
Outbound dial-peers
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

20. SNR is also known as?
...
Debug isdn q931
#voice service voip| #no supplementary-service h225-notify cid-update
Mobile Connect

21. When do you do digit manipulation at the route pattern as opposed to the route list?

No POTS dial-peer or no direct-inward-dial on a pots dial-peer
When multiple sites use the same route pattern - do your digit manipulation on a route list.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

22. How do you enable AAR?
Dtmf-relay h245-alpha
Call-manager-fallback or telephony-service mode 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
SERVICE PARAMETER> Automated Alternate Routing Enable > True

23. What is call-forward pattern used for?
Call forwarding between voip to voip (when CUBE is in play)
Via the voice hunt-group parallel command
(config)#sh cdp neigh detail
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.

24. What is Phone NTP Reference used for?
To notify SIP Phones of NTP
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Dtmf-relay h245-alpha
When they are explicitly matched in a destination-pattern in a dial-peer.

25. On an MGCP GW - how could you see the primary and backup CUCM servers?
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

Show ccm-manager

26. What are the base telephony-service commands for CME?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
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 .)
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
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

27. On the RDP What is the CSS used for?
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.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Mobile Voice Access
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. What are the steps to integrate CUPS with CUCM?
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.
The name configuration field in ephone-dn and voice register dn
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
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

29. 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
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 ....
Assign SIP Dial Rules
...
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

30. How do you configure a gateway to register with gatekeeper?
#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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Automatically configures the MGCP GW for you.

31. What are the commands to manually configure an MGCP gateway?
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
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
#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

32. Name 2 commands to verify RSVP 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)
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.
Single Number Reach (Mobile Connect)
Sh ip rsvp reservation||sh sccp connections

33. With the gw-priority command - does higher or lower priority take precedence?
Higher
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
#after-hours block pattern
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

34. How is NTP sych setup in CUCM?
CUCM OS Administration Settings --> NTP Servers
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
(config)#voice register dialplan
Re-enter the mgcp bind control and mgcp bind media commands.

35. How do you configure AAR?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
#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
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
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

36. What is the Enable Enterprise Feature Access Service Parameter used for?
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 ....
Debug isdn q931
To enable two-stage dialing.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

37. How do you configure an MGCP GW? (router side)
The name configuration field in ephone-dn and voice register dn
(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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Device Pool Locations

38. What is the full E164 format?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
1 ... this is not optional!

39. 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
The Called/Calling Transformations are superceded.
...
(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
CUCM Service Parameters --> Call Park Reversion Timer

40. 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.
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

41. '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
Sh ip rsvp reservation||sh sccp connections
...
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Ccm-manager music-on-hold

42. How do you use an ephone template?
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 .)
Call Simulator. You can use to validate path from router to the PSTN.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Apply it to the ephone.

43. What should you configure before entering auto qos?
Sh ccm-manager
In the CUCM 7.0.1 Release Notes ... search for restore.
Bandwidth
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

44. How do you prioritize route groups?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Put them in a route list.
SIP Route Pattern over a SIP Trunk.

45. An MGCP gateway serving as an SRST router requires what?
...
(config)#sh cdp neigh detail
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Call forwarding between voip to voip (when CUBE is in play)

46. When using drop-through-option What is the max number of huntgroups?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
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.
1 ... this is not optional!

47. Describe how you configure SIP URI functionality.
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

TRUE
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

48. What are the commands to configure a MGCP Gateway? (router)
Device Pool Locations
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
It's best to strip digits at the voice port.
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

49. Is the order of the MRGs in an MRGL significant?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
SIP Route Pattern over a SIP Trunk.

Debug isdn q931

50. How do you setup AutoRegistration in CUCM?
Sends the Calling Name.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
(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

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CCIE Voice Test

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