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. How would you verify that DHCP is working in IOS?
Bottom up - to prevent glare.
Debug isdn q931
(config)#sh cdp neigh detail
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

2. What's the difference between AAR and SRST?
#after-hours block pattern
Sh ip rsvp reservation||sh sccp connections
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
voice translation-profile|and|voice translation-rule

3. What is the bit rate for a G.729 call excluding layer 2?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
...
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.
8kb/s

4. 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?
Sh ccm-manager
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.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Debug isdn q931

5. When a CUCM device dials a number - what happens?
#voice service voip| #no supplementary-service h225-notify cid-update
Show ccm-manager
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 .)
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.

6. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?
Single Number Reach (Mobile Connect)
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
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
Call forwarding between voip to voip (when CUBE is in play)

7. How do you use the # as a string terminator within a SIP Dial Rule?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Debug isdn q931
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Precede the # with a > ... so 9011*>#

8. How do you configure a gateway to register with gatekeeper?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
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
#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

9. When do you do digit manipulation at the route pattern as opposed to the route list?
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
Sh ip rsvp reservation||sh sccp connections
Service Parameter --> Block OffNet To OffNet Transfer --> True

10. What is the Service Parameter to allow the phone that is picking ip to auto answer?
(config)#voice register dialplan
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
voice register pool|and|voice register dn
Auto Call Pickup Enabled

11. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
Show ccm-manager
The Called/Calling Transformations are superceded.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
...

12. What does an H323 GW require that MGCP GWs do not?
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
SIP Route Pattern over a SIP Trunk.
Dial-peers

13. 'If you want to make changes to any softkeys where do you do it?
Allows you to transfer by only pressing the Transfer button once.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

14. What is SIP URI?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
#test voice translation rule 1 <input to test>
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

15. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
voice register pool|and|voice register dn
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Call-manager-fallback or telephony-service mode SRST
SIP Dial Rules

16. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Debug ephone moh
Bottom up - to prevent glare.

17. How much bandwidth does a G.729 call including layer 3 require?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
...
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
24kb/s

18. How do you allow the Calling Name to be sent to the PSTN on a router?
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
voice translation-profile|and|voice translation-rule
Bottom up - to prevent glare.

19. Calls coming from CUCM to PSTN need what?
Outbound dial-peers
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.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

20. What are the 3 mandatory commands within call-manager-fallback?
Max-dn|max-ephone|ip source-address
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
1 ... this is not optional!
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.

21. 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?
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
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)
SERVICE PARAMETER> Automated Alternate Routing Enable > True

22. 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.
To notify SIP Phones of NTP
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Ccm-manager music-on-hold

23. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

24. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
#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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
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

25. Aside from the Media Resources --> Mobile Voice Access where else do you need to specify the the Mobile Voice Access DN?
Service Parameters --> Mobile Voice Access Number

To notify SIP Phones of NTP
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

26. When do you use translate called? Translate calling?
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)#sh cdp neigh detail
(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

27. How do you set up redundancy on outbound dial-peers on an H323 gateway?
#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
To notify SIP Phones of NTP
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

28. What are the commands to configure NTP in IOS?
#voice service voip| #no supplementary-service h225-notify cid-update
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
#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
Debug isdn q931

29. How do you configure SRST?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Call-manager-fallback or telephony-service mode SRST
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
Precede the # with a > ... so 9011*>#

30. How do you get around relying on DNS for your CUCMs?
Dtmf-relay h245-alpha
Call-manager-fallback or telephony-service mode SRST
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

31. How do you use an ephone template?
Apply it to the ephone.
Show ccm-manager
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
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

32. 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
Ccm-manager music-on-hold
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.
SIP Route Pattern over a SIP Trunk.

33. How do you block calls under call-manager-fallback?
Reroute when here is WAN congestion.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
#after-hours block pattern

34. What types of digit manipulation can you perform at the route pattern?
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 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
Matches any length dialed number and truncates it to 4 digits.
Dtmf-relay h245-alpha

35. How do you enable AAR system wide?
Call-manager-fallback or telephony-service mode SRST
Assign SIP Dial Rules
At the CLI: utils ntp status
CUCM Service Parameters --> Automated Alternate Routing Enable --> True

36. How do you prevent toll fraud on CME?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
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 ....

37. What should you configure before entering auto qos?
Call forwarding between voip to voip (when CUBE is in play)
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.
Bandwidth
#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

38. What does ccm-manager config server [IP]</ip> do?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
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
Automatically configures the MGCP GW for you.

39. 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.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
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

40. SNR is also known as?
Allows you to transfer by only pressing the Transfer button once.
The Called/Calling Transformations are superceded.
Mobile Connect
Master/Slave relationship. CUCM controls it.

41. How do you change modes in voice register global?
Call Simulator. You can use to validate path from router to the PSTN.
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Device Pool Locations
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

42. How would you enable security on a GK?
Auto Call Pickup Enabled
...
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie

43. What dtmf-relay type do you use for an H323 GW?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
To enable two-stage dialing.
SIP Dial Rules
Dtmf-relay h245-alpha

44. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?

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45. 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
Allows you to transfer by only pressing the Transfer button once.
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 .)
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

46. 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
Assign a logout-profile to the ephone.
...
Device Pool Locations
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

47. How do you prevent toll fraud on CUCM?
...
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
IP Voice Media Streaming App
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

48. What are the basic ephone-dn and ephone commands?
Apply voice-class codec
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
'Service Parameters --> Auto Answer Timer
(config)#service dhcp|(config)#no ip dhcp conflict logging|(config)#ip dhcp excluded-address <low IP> <hi IP>|(config)#ip dhcp-pool <name>|(dhcpconfig)#network <network name> </nn or mask>|(dhcpconfig)#domain-name <domain>|(dhcpconfig)#dns-server <IP

49. What are the commands to configure SIP phones in CME?
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
24kb/s
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

50. How do you verify that NTP is working on the CUCM server?
Allows you to transfer by only pressing the Transfer button once.
At the CLI: utils ntp status
#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 service voip| #no supplementary-service h225-notify cid-update