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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 CUCM services should you activate?
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
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.

2. When do you do digit manipulation at the route pattern as opposed to the route list?
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
Dial-peers
...
When multiple sites use the same route pattern - do your digit manipulation on a route list.

3. What are the steps to integrate CUPS with CUCM?
#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
The Called/Calling Transformations are superceded.
SIP Dial Rules
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. How do you place SCCP and SIP phones into a single huntgroup?
'Service Parameters --> Auto Answer Timer
Bottom up - to prevent glare.
#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
Via the voice hunt-group parallel command

5. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
Mobile Connect
#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
#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
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.

6. How would you verify that DHCP is working in IOS?

(config)#sh cdp neigh detail
Matches any length dialed number and truncates it to 4 digits.
Bandwidth

7. What must you do for a BACD script to work on a CME router?

CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
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
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au

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

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9. What do you need to do to activate the CME GUI?
#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
Max-dn|max-ephone|ip source-address
#after-hours block pattern
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

10. 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 ....
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

11. What are the 3 mandatory commands within call-manager-fallback?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Debug isdn q931
Outbound dial-peers
Max-dn|max-ephone|ip source-address

12. What are the commands to create vlans on an ESW?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
When configuring TEHO.
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
CUCM Service Parameters --> Call Park Reversion Timer

13. When can you not use a Standard Local Route Group?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
sh ephone reg|and|debug voip dialpeer
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
When configuring TEHO.

14. 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)
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
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
Put them in a route list.

15. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
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
When they are explicitly matched in a destination-pattern in a dial-peer.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

16. How do you set the inter-digit timeout for SIP phones in CME?
#after-hours block pattern
'Service Parameters --> Auto Answer Timer
#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
(config)#voice register dialplan

17. What are the commands to configure NTP in IOS?
On CUCM it's identical to adding an H323 GW.
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
#test voice translation rule 1 <input to test>
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

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
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
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 ....
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

19. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
Sends the Calling Name.
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 .)
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable

20. What CM Service needs to be start in Serviceability for MOH to work?
IP Voice Media Streaming App
Debug isdn q931
#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
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify

21. If you're using Called/Calling Party Transformations at the Route Pattern and tranformations at the route list ... What does the RP affect and What does the RL affect?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Reroute when here is WAN congestion.
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.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

22. What are the commands to configure SRST in fallback?
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
Call-manager-fallback or telephony-service mode SRST
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

23. When are digits stripped in a gateway?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
When they are explicitly matched in a destination-pattern in a dial-peer.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files

24. How do you verify that NTP is working on the CUCM server?
Assign a logout-profile to the ephone.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
At the CLI: utils ntp status

25. How do you create a trunk on the router side?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Apply voice-class codec
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

26. 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
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
#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
...

27. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
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
'Use before the ... so XXXX
Bottom up - to prevent glare.
At the CLI: utils ntp status

28. What's the difference between AAR and SRST?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
In CUCM - configure CFUR to point to it's E164 number.
When configuring TEHO.

29. What does an H323 GW require that MGCP GWs do not?

Dial-peers
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)
When they are explicitly matched in a destination-pattern in a dial-peer.

30. Name 4 useful show commands for active calls.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
...
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
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

31. How do you use the # as a string terminator within a SIP Dial Rule?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Precede the # with a > ... so 9011*>#
1 ... this is not optional!
SIP Dial Rules

32. On the RDP What is the CSS used for?
#test voice translation rule 1 <input to test>
Mobile Voice Access
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)
Reroute when there is a WAN Outage.

33. Where do you use VIA zone?
Assign a logout-profile to the ephone.
Dial-peers
Phone number followed by domain name.|i.e. 3006@ipxcme.com
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

34. An MGCP gateway serving as an SRST router requires what?
...
Auto Call Pickup Enabled
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Redundant dial plan. One on CUCM and one on the routers via dial-peers.

35. How do you create a trunk on the switch side?
SERVICE PARAMETER> Automated Alternate Routing Enable > 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.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

36. What are the basic ephone-dn and ephone commands?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
In CUCM - configure CFUR to point to it's E164 number.

37. In CME Where is the Calling Name derived from?
To notify SIP Phones of NTP
The name configuration field in ephone-dn and voice register dn
Call forwarding between voip to voip (when CUBE is in play)
24kb/s

38. What's the best way to perform digit manipulation on a route group?
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
Configure it on the route group through the route list - then it will be local to the route list.
In the CUCM 7.0.1 Release Notes ... search for restore.
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

39. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
To enable two-stage dialing.
SIP Dial Rules
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

40. 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
Master/Slave relationship. CUCM controls it.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Outbound dial-peers

41. 'How do you inform a SIP phone of NTP information?
Higher
#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 Simulator. You can use to validate path from router to the PSTN.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

42. What is SIP URI?
Sh ip rsvp reservation||sh sccp connections
Phone number followed by domain name.|i.e. 3006@ipxcme.com
CUCM OS Administration Settings --> NTP Servers
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 .)

43. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
TRUE
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
When configuring TEHO.

44. What is the bit rate for a G.729 call excluding layer 2?
At the CLI: utils ntp status
Call forwarding between voip to voip (when CUBE is in play)
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
8kb/s

45. What commands are needed to configure the voice register pool in CME?
Single Number Reach (Mobile Connect)
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.
TRUE
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

46. How do you verify where MOH is being served up from?

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47. Which takes precedence Device Locations or Device Pool Locations?
Device Pool Locations
On CUCM it's identical to adding an H323 GW.
The name configuration field in ephone-dn and voice register dn
Apply it to the ephone.

48. How can you confirm the MGCP GW is registered to CUCM - in IOS?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Sh ccm-manager
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

49. How do you change modes in voice register global?
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
voice register pool|and|voice register dn
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

50. How do you enable AAR system wide?
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
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Configure it on the route group through the route list - then it will be local to the route list.