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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. 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
...
Device Pool Locations
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

2. '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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
...
Bottom up - to prevent glare.

3. How do you 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
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.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

4. When using drop-through-option What is the max number of huntgroups?
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
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.

5. What are the commands to configure a SIP phone in CUCME?
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.
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 Route Pattern over a SIP Trunk.
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

6. How do you see the details of calls coming in and out of the PRI?
To enable two-stage dialing.
Debug isdn q931
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Reroute when there is a WAN Outage.

7. How do you see multicast packets being sent?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Debug ephone moh
Service Parameters --> Mobile Voice Access Number
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.

8. 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?
Call forwarding between voip to voip (when CUBE is in play)
(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
Automatically configures the MGCP GW for you.
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)

9. What is the bit rate for a G.729 call excluding layer 2?
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.
8kb/s
Allows you to transfer by only pressing the Transfer button once.
(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

10. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
SIP Dial Rules
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
At the CLI: utils ntp status
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

11. How would you enable security on a GK?
In CUCM - configure CFUR to point to it's E164 number.
Outbound dial-peers
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

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

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13. Describe how you configure SIP URI functionality.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
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

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

14. What dtmf-relay type do you use for an H323 GW?
SIP Dial Rules
Dtmf-relay h245-alpha
(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
...

15. How do you block an external call from being transferred back out to the pstn by an internal user?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Reroute when here is WAN congestion.
Service Parameter --> Block OffNet To OffNet Transfer --> True
24kb/s

16. What are the commands to configure an H323 GW?
Apply it to the ephone.
In CUCM - configure CFUR to point to it's E164 number.
#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
Mobile Connect

17. An MGCP gateway serving as an SRST router requires what?
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
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
#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
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

18. How do you enable Extension Mobility for a device in CME?
Precede the # with a > ... so 9011*>#
Assign a logout-profile to the ephone.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

19. What is a best practice for digit manipulation - in regards to H323 GWs?
#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
Call Simulator. You can use to validate path from router to the PSTN.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Device Pool Locations

20. What do you need to do to activate the CME GUI?
When configuring TEHO.
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
Allows you to transfer by only pressing the Transfer button once.
Device Pool Locations

21. How do do configure TEHO?
Precede the # with a > ... so 9011*>#
At the CLI: utils ntp status
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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

22. What are the commands to configure SIP phones in CME?
#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
SIP Route Pattern over a SIP Trunk.
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
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

23. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
#voice service voip| #no supplementary-service h225-notify cid-update
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer

24. How do you prioritize route groups?
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Re-enter the mgcp bind control and mgcp bind media commands.
Put them in a route list.

25. 'How do you inform a SIP phone of NTP information?
When they are explicitly matched in a destination-pattern in a dial-peer.
In the CUCM 7.0.1 Release Notes ... search for restore.
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.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

26. How do you prevent toll fraud on CME?
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 ....
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Mobile Voice Access
Higher

27. What's the difference between AAR and SRST?
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
#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
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

28. How do you set the Call Park Reversion Timer?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
CUCM Service Parameters --> Call Park Reversion Timer
Sh ccm-manager
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)

29. What are the commands to create vlans on an ESW?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
SIP Route Pattern over a SIP Trunk.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Allows you to transfer by only pressing the Transfer button once.

30. How do you allow H323 calls to be preserved should the primary H323 GW fail?
#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
Dial-peers
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

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

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32. What is SIP URI?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
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
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.
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

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

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34. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
Bottom up - to prevent glare.
Higher
#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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

35. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
#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
SIP Route Pattern over a SIP Trunk.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

36. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?
SIP Route Pattern over a SIP Trunk.
(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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files

37. In Gatekeeper CAC how do you restrict a specific endpoint?
IP Voice Media Streaming App
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
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.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

38. When do you use translate called? Translate calling?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

39. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Single Number Reach (Mobile Connect)
Re-enter the mgcp bind control and mgcp bind media commands.

40. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
Ccm-manager music-on-hold
Bandwidth

Redundant dial plan. One on CUCM and one on the routers via dial-peers.

41. What are two caveats to using the ccm-manager config server</ip> command?
(config)#voice register dialplan
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True

42. What CUCM services should you activate?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
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
(config)#sh cdp neigh detail

43. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Mobile Connect
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
When configuring TEHO.

44. 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.
SIP Route Pattern over a SIP Trunk.
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable

45. Describe the relationship between route patterns and end devices.
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.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Sh ip rsvp reservation||sh sccp connections

46. How would you verify that DHCP is working in IOS?
#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
Mobile Voice Access
(config)#sh cdp neigh detail
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

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

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48. What commands are needed to configure the voice register pool in CME?
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.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
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

49. 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.
IP Voice Media Streaming App
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
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

50. On the RDP What is the Rerouting CSS used for?
Dial-peers
SIP Dial Rules
Single Number Reach (Mobile Connect)
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast