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. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
'Use before the ... so XXXX
(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
To notify SIP Phones of NTP
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

2. What's the best way to do digit manipulation on an IOS gateway?
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
voice translation-profile|and|voice translation-rule
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

3. What would force you to use telephony-service to configure SRST?
Configure it on the route group through the route list - then it will be local to the route list.
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 .)
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

4. How do do configure TEHO?
Bandwidth
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
#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
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!

5. How do you configure AAR?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
SIP Route Pattern over a SIP Trunk.

6. How much bandwidth does a G.729 call including layer 3 require?
To notify SIP Phones of NTP
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
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
24kb/s

7. What are the base telephony-service commands for CME?
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
SIP Dial Rules
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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

8. 'If you're not getting a DHCP address from CUCM what then?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
...
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
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

9. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
When configuring TEHO.
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)
SIP Route Pattern over a SIP Trunk.
Call-manager-fallback or telephony-service mode SRST

10. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
Ccm-manager music-on-hold
...
In the CUCM 7.0.1 Release Notes ... search for restore.
voice register pool|and|voice register dn

11. What are the steps to integrate CUPS with CUCM?
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
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
Precede the # with a > ... so 9011*>#
To enable two-stage dialing.

12. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
Dtmf-relay h245-alpha
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
SIP Dial Rules

13. How is NTP sych setup in CUCM?
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.
CUCM OS Administration Settings --> NTP Servers
Sh ccm-manager
#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

14. How do you configure phone ports on a 3750?
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
(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
The Called/Calling Transformations are superceded.
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST

15. What's the best way to perform digit manipulation on a route group?
Bottom up - to prevent glare.
Configure it on the route group through the route list - then it will be local to the route list.
IP Voice Media Streaming App
Phone number followed by domain name.|i.e. 3006@ipxcme.com

16. What are the commands to configure SRST in fallback?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
(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
Reroute when there is a WAN Outage.

17. How do you ensure that G.711 only is used?
Automatically configures the MGCP GW for you.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable

Via the voice hunt-group parallel command

18. How do you prioritize route groups?
Ccm-manager music-on-hold
Put them in a route list.
Call forwarding between voip to voip (when CUBE is in play)
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

19. What are the commands to manually configure an MGCP gateway?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
#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
IP Voice Media Streaming App
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

20. What are the commands to configure NTP in IOS?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
(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
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
Apply it to the ephone.

21. What are the basic ephone-dn and ephone commands?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
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
#voice service voip| #no supplementary-service h225-notify cid-update

22. When can you not use a Standard Local Route Group?
(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
When configuring TEHO.
The Called/Calling Transformations are superceded.
Precede the # with a > ... so 9011*>#

23. When are digits stripped in a gateway?
#test voice translation rule 1 <input to test>
When they are explicitly matched in a destination-pattern in a dial-peer.
(config)#sh cdp neigh detail
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie

24. What does ccm-manager config server [IP]</ip> do?
Automatically configures the MGCP GW for you.
#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
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
Mobile Voice Access

25. How do you support multiple codecs on a dialpeer?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Apply voice-class codec
(config)#sh cdp neigh detail
The name configuration field in ephone-dn and voice register dn

26. What are the commands to configure a MGCP Gateway? (router)
On CUCM it's identical to adding an H323 GW.
Configure it on the route group through the route list - then it will be local to the route list.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
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. Name 4 useful show commands for active calls.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
#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
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

28. What is Phone NTP Reference used for?
To notify SIP Phones of NTP
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.

29. What terminology translates to SRST?
Reroute when there is a WAN Outage.
#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
'Use before the ... so XXXX
(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

30. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Ccm-manager music-on-hold
Outbound dial-peers
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 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

31. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
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 ipv4:10.1.5.3 (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

32. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Re-enter the mgcp bind control and mgcp bind media commands.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
...
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.

33. How do you setup AutoRegistration in CUCM?
TRUE
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
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

34. How do you enable AAR system wide?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
voice register pool|and|voice register dn
Higher
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

35. 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

36. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Put them in a route list.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

37. On an MGCP GW - how could you see the primary and backup CUCM servers?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Re-enter the mgcp bind control and mgcp bind media commands.
Show ccm-manager
Ccm-manager music-on-hold

38. What types of digit manipulation can you perform at the route pattern?
Debug ephone moh
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

39. How do you set up redundancy on outbound dial-peers on an H323 gateway?
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
Debug ephone moh
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)

40. What are the steps to configure Single Number Reach and Mobile Voice Access?
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.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer

41. 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

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

42. What is Mobile Voice Access?
Single Number Reach (Mobile Connect)
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Call-manager-fallback or telephony-service mode SRST
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

43. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Device Pool Locations
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

44. How do you use an ephone template?
Automatically configures the MGCP GW for you.
Sh ip rsvp reservation||sh sccp connections
Apply it to the ephone.
In the CUCM 7.0.1 Release Notes ... search for restore.

45. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
#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
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

46. Does CUCM support RSVP natively?
It's best to strip digits at the voice port.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
'Use before the ... so XXXX
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

47. How do you prevent H323 caller-id updates to CUCM
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Put them in a route list.
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

48. How do you see multicast packets being sent?
Debug ephone moh
Service Parameter --> Block OffNet To OffNet Transfer --> True
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Sh ccm-manager

49. How do you configure SRST?
Call-manager-fallback or telephony-service mode SRST
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

50. What are the commands to configure a SIP phone in CUCME?
Assign a logout-profile to the ephone.
#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
At the CLI: utils ntp status
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