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CCIE Voice Test

Instructions:

Answer 50 questions in 15 minutes.
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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 do you configure SIP phones to be able to call other SIP phones via SIP URI?
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 ....
Ccm-manager music-on-hold
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
SIP Route Pattern over a SIP Trunk.

2. How do you configure CUBE?

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3. How do you prevent toll fraud on CUCM?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
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

4. What is CSIM?
Call Simulator. You can use to validate path from router to the PSTN.

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.
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

5. 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
(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
SIP Route Pattern over a SIP Trunk.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

6. What is Phone NTP Reference used for?
To notify SIP Phones of NTP
Master/Slave relationship. CUCM controls it.
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
Single Number Reach (Mobile Connect)

7. 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
Assign a logout-profile to the ephone.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
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)

8. What are the basic SCCP Commands fro telephony-service in CUCME?
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
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

9. What are the base telephony-service commands for CME?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
When configuring TEHO.
Auto Call Pickup Enabled

10. 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.
Sends the Calling Name.
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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

11. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
SIP Dial Rules
Via the voice hunt-group parallel command
Master/Slave relationship. CUCM controls it.
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

12. How do you prioritize route groups?
Put them in a route list.
voice translation-profile|and|voice translation-rule
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Top Down means that channel 1 will be the first channel used to place outgoing calls.

13. How do you configure class of service (CoS) in CUCM? CME?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Call-manager-fallback or telephony-service mode SRST

14. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Master/Slave relationship. CUCM controls it.
Bottom up - to prevent glare.
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
Bandwidth

15. Name 2 commands to verify RSVP functionality.
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
Apply voice-class codec
Sh ip rsvp reservation||sh sccp connections

16. When can you not use a Standard Local Route 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)
When configuring TEHO.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

17. Where can you assign the AAR Group?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Call-manager-fallback or telephony-service mode SRST
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.

18. What terminology translates to SRST?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Apply it to the ephone.
#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
Reroute when there is a WAN Outage.

19. 'How do you inform a SIP phone of NTP information?
Max-dn|max-ephone|ip source-address
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

20. How do you disable KPML?
Configure it on the route group through the route list - then it will be local to the route list.
Assign SIP Dial Rules
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
When they are explicitly matched in a destination-pattern in a dial-peer.

21. How do you prevent H323 caller-id updates to CUCM
#voice service voip| #no supplementary-service h225-notify cid-update
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
Auto Call Pickup Enabled
Apply it to the ephone.

22. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Call-manager-fallback or telephony-service mode SRST
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

23. What are the 3 mandatory commands within call-manager-fallback?
Reroute when here is WAN congestion.

Max-dn|max-ephone|ip source-address
SIP Route Pattern over a SIP Trunk.

24. 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?
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)

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
Reroute when here is WAN congestion.

25. Two useful troubleshooting commands for CUCME?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
sh ephone reg|and|debug voip dialpeer

26. What dtmf-relay type do you use for an H323 GW?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Dtmf-relay h245-alpha
Debug ephone moh
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

27. What does an H323 GW require that MGCP GWs do not?
CUCM OS Administration Settings --> NTP Servers
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Via the voice hunt-group parallel command
Dial-peers

28. On the RDP What is the Rerouting CSS used for?
(config)#sh cdp neigh detail
Single Number Reach (Mobile Connect)
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
(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

29. How do you create a trunk on the router side?
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)
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
Debug ephone moh
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files

30. How do you configure CUCM redundancy on an H323 gateway?
It's best to strip digits at the voice port.
#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
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Sh ccm-manager

31. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
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
Re-enter the mgcp bind control and mgcp bind media commands.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True

32. When a CUCM device dials a number - what happens?
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 .)
Bottom up - to prevent glare.
voice register pool|and|voice register dn
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.

33. What does KPML do?
Dial-peers
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
#after-hours block 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 .)

34. How do you set the Call Park Reversion Timer?
Assign SIP Dial Rules
#test voice translation rule 1 <input to test>
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.
CUCM Service Parameters --> Call Park Reversion Timer

35. 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.
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
Apply voice-class codec
Max-dn|max-ephone|ip source-address

36. In Gatekeeper CAC how do you restrict a specific endpoint?
Mobile Connect
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Call forwarding between voip to voip (when CUBE is in play)
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

37. How do you enable AAR?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Dtmf-relay h245-alpha
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

38. What's the best way to perform digit manipulation on a route group?
...
Apply voice-class codec
Configure it on the route group through the route list - then it will be local to the route list.
The Called/Calling Transformations are superceded.

39. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
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
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.

40. How do you configure DHCP in IOS?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Apply it to the ephone.
sh ephone reg|and|debug voip dialpeer
(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

41. What does rule 1 /^.*(....$)/ /1/ do?
At the CLI: utils ntp status
Higher
Matches any length dialed number and truncates it to 4 digits.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.

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

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43. 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-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

44. 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
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
(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
In CUCM - configure CFUR to point to it's E164 number.

45. When a device is roaming to a different site How does it get the Roaming Sensitive Settings from the Device Pool?
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.
sh ephone reg|and|debug voip dialpeer
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Max-dn|max-ephone|ip source-address

46. What should you configure before entering auto qos?
Call Simulator. You can use to validate path from router to the PSTN.
Bandwidth
Precede the # with a > ... so 9011*>#
Single Number Reach (Mobile Connect)

47. Calls coming from CUCM to PSTN need what?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Outbound dial-peers
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
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

48. Where is it best to manipulate digits for inbound calls?

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49. What's the difference between AAR and SRST?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Put them in a route list.
Debug isdn q931
Show ccm-manager

50. What does ccm-manager config server [IP]</ip> do?
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
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
(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.