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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. What CM Service needs to be start in Serviceability for MOH to work?
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
IP Voice Media Streaming App
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
SIP Dial Rules

2. When can you not use a Standard Local Route Group?
When configuring TEHO.
When they are explicitly matched in a destination-pattern in a dial-peer.
The Called/Calling Transformations are superceded.
In CUCM - configure CFUR to point to it's E164 number.

3. On an H323 GW - how do you adjust the timers for redundancy hunting?
Apply voice-class codec
Ccm-manager music-on-hold
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
(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

4. Privacy is enabled system-wide in CUCM by default. (T or F)
Top Down means that channel 1 will be the first channel used to place outgoing calls.
TRUE
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

5. What's the best way to do digit manipulation on an IOS gateway?
Reroute when there is a WAN Outage.
voice translation-profile|and|voice translation-rule
Call forwarding between voip to voip (when CUBE is in play)
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

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

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7. What is the full E164 format?
Single Number Reach (Mobile Connect)
Apply voice-class codec
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

8. What does an H323 GW require that MGCP GWs do not?
#test voice translation rule 1 <input to test>
(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
Dtmf-relay h245-alpha
Dial-peers

9. 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.
Assign SIP Dial Rules
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.

10. How can you confirm the MGCP GW is registered to CUCM - in IOS?
Sh ccm-manager

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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

11. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Ccm-manager music-on-hold
SIP Route Pattern over a SIP Trunk.
...

12. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
Re-enter the mgcp bind control and mgcp bind media commands.
SIP Route Pattern over a SIP Trunk.

13. 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
TRUE
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie

14. What is CSIM?
Matches any length dialed number and truncates it to 4 digits.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Call Simulator. You can use to validate path from router to the PSTN.

15. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?

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16. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
SIP Route Pattern over a SIP Trunk.
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
In the CUCM 7.0.1 Release Notes ... search for restore.
In CUCM - configure CFUR to point to it's E164 number.

17. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
24kb/s
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
CUCM OS Administration Settings --> NTP Servers
Redundant dial plan. One on CUCM and one on the routers via dial-peers.

18. What types of digit manipulation can you perform at the route pattern?
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 .)
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
Ccm-manager music-on-hold
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!

19. How do you support multiple codecs on a dialpeer?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Apply voice-class codec
SIP Route Pattern over a SIP Trunk.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!

20. Is the order of the MRGs in an MRGL significant?
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
It's best to strip digits at the voice port.
'Service Parameters --> Auto Answer Timer
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.

21. Calls coming from CUCM to PSTN need what?
At the CLI: utils ntp status
Outbound dial-peers
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
SERVICE PARAMETER> Automated Alternate Routing Enable > True

22. Name 4 useful show commands for active calls.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
#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
Sh ccm-manager

23. What terminology translates to AAR?
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.
voice register pool|and|voice register dn
Reroute when here is WAN congestion.
CUCM OS Administration Settings --> NTP Servers

24. How do you see multicast packets being sent?
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
#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

Debug ephone moh

25. What CUCM services should you activate?
Assign a logout-profile to the ephone.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Re-enter the mgcp bind control and mgcp bind media commands.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

26. Describe the relationship between route patterns and end devices.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Configure it on the route group through the route list - then it will be local to the route list.
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.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

27. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
The Called/Calling Transformations are superceded.
'Service Parameters --> Auto Answer Timer
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 .)
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

28. What do you need to do to activate the CME GUI?
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 -- UPTSAL||Create Your Users|Phone - set owner id|Create SIP Trunk|SIP Trunk Security Profile||Application Server|Licensing||||On CUPS||Work Left to right
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

29. How do you block calls under call-manager-fallback?
#after-hours block pattern
When configuring TEHO.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.

30. How would you enable security on a GK?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Call-manager-fallback or telephony-service mode SRST
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

31. How do you place SCCP and SIP phones into a single huntgroup?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Call Simulator. You can use to validate path from router to the PSTN.

Via the voice hunt-group parallel command

32. Name 2 commands to verify RSVP functionality.
#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
Mobile Connect
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Sh ip rsvp reservation||sh sccp connections

33. What are the commands to configure a SIP phone in CUCME?
Assign 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
It's best to strip digits at the voice port.
Call Simulator. You can use to validate path from router to the PSTN.

34. SNR is also known as?
Mobile Connect
Put them in a route list.
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
Mobile Voice Access

35. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
Service Parameters --> Mobile Voice Access Number

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.
Allows you to transfer by only pressing the Transfer button once.

36. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
sh ephone reg|and|debug voip dialpeer

No POTS dial-peer or no direct-inward-dial on a pots dial-peer

37. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
(config)#sh cdp neigh detail
#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
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

38. On the RDP What is the Rerouting CSS used for?
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
Single Number Reach (Mobile Connect)
Max-dn|max-ephone|ip source-address
Service Parameter --> Block OffNet To OffNet Transfer --> True

39. How is NTP sych setup in CUCM?
CUCM OS Administration Settings --> NTP Servers
At the CLI: utils ntp status
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.
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 .)

40. What dtmf-relay type do you use for an H323 GW?
Dtmf-relay h245-alpha
#after-hours block pattern
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
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

41. What are the basic SCCP Commands fro telephony-service in CUCME?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
(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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

42. What is a best practice for digit manipulation - in regards to H323 GWs?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Matches any length dialed number and truncates it to 4 digits.
Service Parameter --> Block OffNet To OffNet Transfer --> True
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

43. What does the Transfer On-hook Enabled Service Parameter do?
When they are explicitly matched in a destination-pattern in a dial-peer.
In the CUCM 7.0.1 Release Notes ... search for restore.
Allows you to transfer by only pressing the Transfer button once.
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

44. Where do you use VIA zone?
Mobile Voice Access
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
The name configuration field in ephone-dn and voice register dn
sh ephone reg|and|debug voip dialpeer

45. What are the commands to configure a MGCP Gateway? (router)
The name configuration field in ephone-dn and voice register dn
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
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

46. What is Phone NTP Reference used for?
#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
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
To notify SIP Phones of NTP
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

47. What are the commands to create vlans on an ESW?
The Called/Calling Transformations are superceded.
Call Simulator. You can use to validate path from router to the PSTN.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

48. How would you set the the timer for Auto Answer?

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49. How do you allow the Calling Name to be sent to the PSTN on a router?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
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 .)

50. How do do configure TEHO?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Device Pool Locations
#after-hours block pattern
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