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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. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
Assign SIP Dial Rules
To notify SIP Phones of NTP
In the CUCM 7.0.1 Release Notes ... search for restore.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

2. How do you allow H323 calls to be preserved should the primary H323 GW fail?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
In CUCM - configure CFUR to point to it's E164 number.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Assign SIP Dial Rules

3. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Bottom up - to prevent glare.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Assign SIP Dial Rules

4. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
Outbound dial-peers
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
To notify SIP Phones of NTP
The name configuration field in ephone-dn and voice register dn

5. In CME Where is the Calling Name derived from?
1 ... this is not optional!
The name configuration field in ephone-dn and voice register dn
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-manager-fallback or telephony-service mode SRST

6. How do you place SCCP and SIP phones into a single huntgroup?
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
'Service Parameters --> Auto Answer Timer
Via the voice hunt-group parallel command

7. 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.
Service Parameters --> Mobile Voice Access Number

8. How do you change modes in voice register global?
Bottom up - to prevent glare.
#after-hours block pattern
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
1 ... this is not optional!

9. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
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
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
SIP Dial Rules

10. Describe how you configure SIP URI functionality.
1 ... this is not optional!
(config)#voice register dialplan
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

11. How do you use an ephone template?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Apply it to the ephone.
24kb/s

12. What are the base telephony-service commands for CME?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

13. 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
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
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
...
'Use before the ... so XXXX

14. How do you set up redundancy on outbound dial-peers on an H323 gateway?
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.
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
...
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

15. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Sh ccm-manager
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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

16. What must you do for a BACD script to work on a CME router?
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
Single Number Reach (Mobile Connect)
CUCM Service Parameters --> Call Park Reversion Timer
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

17. How do you set the inter-digit timeout for SIP phones in CME?
(config)#voice register dialplan
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Precede the # with a > ... so 9011*>#
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

18. 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.
Via the voice hunt-group parallel command
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

19. How do do configure TEHO?
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
Put them in a route list.
...
Debug ephone moh

20. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?

SIP Route Pattern over a SIP Trunk.
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.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

21. Name 2 commands to verify RSVP functionality.
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
Call forwarding between voip to voip (when CUBE is in play)
Sh ip rsvp reservation||sh sccp connections
#after-hours block pattern

22. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
At the CLI: utils ntp status
(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
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files

23. What is call-forward pattern used for?
Call forwarding between voip to voip (when CUBE is in play)
Reroute when here is WAN congestion.

Configure it on the route group through the route list - then it will be local to the route list.

24. 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?
Bandwidth
(config)#voice register dialplan
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
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.

25. When are digits stripped in a gateway?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

When they are explicitly matched in a destination-pattern in a dial-peer.
Assign SIP Dial Rules

26. Maximum Wait Time for Desk Pickup?
Dtmf-relay h245-alpha
Int vlan 130|ip address 10.10.101.1 255.255.255.0|int vlan 240|ip address 10.10.201.1 255.255.255.0
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

27. When setting up SIP URI where do you configure the CUCM's domain name?'

#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
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
CUCM OS Administration Settings --> NTP Servers

28. How do you configure CUBE?

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29. What commands are needed to configure the voice register pool in CME?
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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

30. How do you see the details of calls coming in and out of the PRI?
...
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Debug isdn q931
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

31. What are the commands to create the L3 routing interface for VLANS (SVI)?
Show ccm-manager
Int vlan 130|ip address 10.10.101.1 255.255.255.0|int vlan 240|ip address 10.10.201.1 255.255.255.0
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 .)
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

32. On an H323 GW - how do you adjust the timers for redundancy hunting?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
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 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
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

33. How do you configure DHCP in IOS?
(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
Mobile Connect
Auto Call Pickup Enabled
SIP Dial Rules

34. 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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Single Number Reach (Mobile Connect)
Phone number followed by domain name.|i.e. 3006@ipxcme.com

35. What CUCM services should you activate?
Call Simulator. You can use to validate path from router to the PSTN.
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.
#after-hours block pattern
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.

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

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37. How do you prevent H323 caller-id updates to CUCM
#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
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
#voice service voip| #no supplementary-service h225-notify cid-update
(config)#sh cdp neigh detail

38. What does an H323 GW require that MGCP GWs do not?
#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
Mobile Voice Access
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 .)
Dial-peers

39. 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)
At the CLI: utils ntp status
Master/Slave relationship. CUCM controls it.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

40. What are the commands to configure NTP in IOS?

(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
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

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 os truly unsurpassed. I have opened up a wealth of opportunities for myself and my family and
(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
...
Debug isdn q931
Device Pool Locations

42. How do you configure a SIP Trunk? (router side)
Top Down means that channel 1 will be the first channel used to place outgoing calls.
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
(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

43. 'If you want to make changes to any softkeys where do you do it?
voice translation-profile|and|voice translation-rule
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
(config)#voice register dialplan
Call-manager-fallback or telephony-service mode SRST

44. What is Mobile Voice Access?
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
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
(config)#sh cdp neigh detail

45. When using drop-through-option What is the max number of huntgroups?
#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 ... this is not optional!
It's best to strip digits at the voice port.

46. Where can you assign the AAR Group?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
voice register pool|and|voice register dn
Allows you to transfer by only pressing the Transfer button once.
To enable two-stage dialing.

47. What is the full E164 format?
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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
The name configuration field in ephone-dn and voice register dn

48. How do you configure SRST?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Call-manager-fallback or telephony-service mode SRST
1 ... this is not optional!
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

49. Calls coming from CUCM to PSTN need what?

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.
Outbound dial-peers
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True

50. How do you see multicast packets being sent?
Debug ephone moh
Outbound dial-peers
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.
When configuring TEHO.