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

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2. Name 2 commands to verify RSVP functionality.
The name configuration field in ephone-dn and voice register dn
Sh ip rsvp reservation||sh sccp connections
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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.

3. When do you use translate called? Translate calling?
(config)#sh cdp neigh detail
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
Allows you to transfer by only pressing the Transfer button once.

4. What are the steps to configure Single Number Reach and Mobile Voice Access?
'Service Parameters --> Auto Answer Timer

...
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

5. What is CSIM?
(config)#sh cdp neigh detail
Apply voice-class codec
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Call Simulator. You can use to validate path from router to the PSTN.

6. What commands are needed to configure the voice register pool in CME?
It's best to strip digits at the voice port.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
voice register pool|and|voice register 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

7. 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?
Assign SIP Dial Rules
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
(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
...

8. Name 4 useful show commands for active calls.
#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
When configuring TEHO.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

9. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
voice register pool|and|voice register dn
Via the voice hunt-group parallel command
#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

10. When setting up SIP URI where do you configure the CUCM's domain name?'
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
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
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name

11. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Call Simulator. You can use to validate path from router to the PSTN.
Ccm-manager music-on-hold

12. 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.
(config)#sh cdp neigh detail
CUCM OS Administration Settings --> NTP Servers
Ccm-manager music-on-hold

13. On an MGCP GW - how could you see the primary and backup CUCM servers?
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
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
Show ccm-manager
Top Down means that channel 1 will be the first channel used to place outgoing calls.

14. What's the difference between AAR and SRST?
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.
Put them in a route list.
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

15. How do you configure phone ports on an ESW?
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
'Use before the ... so XXXX
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
voice register pool|and|voice register dn

16. How do you configure a SIP Trunk? (router side)
(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
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
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
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

17. What are the commands to create the L3 routing interface for VLANS (SVI)?
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 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
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

18. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
It's best to strip digits at the voice port.
The Called/Calling Transformations are superceded.
Dial-peers

19. How do you allow H323 calls to be preserved should the primary H323 GW fail?
Debug ephone moh
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
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

20. How would you enable security on a GK?
Call Simulator. You can use to validate path from router to the PSTN.
Re-enter the mgcp bind control and mgcp bind media commands.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
SERVICE PARAMETER> Automated Alternate Routing Enable > True

21. On an H323 GW - how do you adjust the timers for redundancy hunting?
Ccm-manager music-on-hold
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
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
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

22. On the RDP What is the Rerouting CSS used for?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Single Number Reach (Mobile Connect)
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

23. 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?
Bottom up - to prevent glare.
Show 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.
Ccm-manager music-on-hold

24. What are the commands to set up the PRI?

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25. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
SIP Route Pattern over a SIP Trunk.
...
When multiple sites use the same route pattern - do your digit manipulation on a route list.
On CUCM it's identical to adding an H323 GW.

26. What does ccm-manager config server [IP]</ip> do?
...
Higher
CUCM OS Administration Settings --> NTP Servers
Automatically configures the MGCP GW for you.

27. What's the best way to do digit manipulation on an IOS gateway?
voice translation-profile|and|voice translation-rule
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
(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
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST

28. How do you block an external call from being transferred back out to the pstn by an internal user?
Single Number Reach (Mobile Connect)
Service Parameter --> Block OffNet To OffNet Transfer --> True
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
...

29. What are the commands to manually configure an MGCP gateway?
'Service Parameters --> Auto Answer Timer
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
#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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

30. How do you verify that NTP is working on the CUCM server?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
At the CLI: utils ntp status
Show ccm-manager
It's best to strip digits at the voice port.

31. What is Phone NTP Reference used for?
To notify SIP Phones of NTP
#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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
#voice service voip| #no supplementary-service h225-notify cid-update

32. With the gw-priority command - does higher or lower priority take precedence?
Single Number Reach (Mobile Connect)
Reroute when there is a WAN Outage.
Higher
...

33. 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
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
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
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

34. How do you change modes in voice register global?
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST

Sends the Calling Name.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.

35. 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
...
Higher
#after-hours block pattern
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit

36. 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

...
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

37. What terminology translates to AAR?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Reroute when here is WAN congestion.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Sends the Calling Name.

38. When a CUCM device dials a number - what happens?
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
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 .)
sh ephone reg|and|debug voip dialpeer
Higher

39. How do you create a trunk on the router side?
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

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
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

40. What are two caveats to using the ccm-manager config server</ip> command?
To enable two-stage dialing.
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 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

41. What are the basic ephone-dn and ephone commands?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
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
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
It's best to strip digits at the voice port.

42. When using drop-through-option What is the max number of huntgroups?
voice register pool|and|voice register dn
1 ... this is not optional!
At the CLI: utils ntp status
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

43. In Gatekeeper CAC how do you restrict a specific endpoint?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

44. What are the commands to configure a SIP phone in CUCME?
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
Single Number Reach (Mobile Connect)
Max-dn|max-ephone|ip source-address
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable

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

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46. 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
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Debug ephone moh
Call-manager-fallback or telephony-service mode SRST

47. How do you block calls under call-manager-fallback?

#after-hours block pattern
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Dial-peers

48. What does rule 1 /^.*(....$)/ /1/ do?
Dtmf-relay h245-alpha
Matches any length dialed number and truncates it to 4 digits.
Assign a logout-profile to the ephone.
...

49. How do you set the Call Park Reversion Timer?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
CUCM Service Parameters --> Call Park Reversion Timer
Device Pool Locations
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

50. Which takes precedence Device Locations or Device Pool Locations?
...
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Device Pool Locations
Service Parameters --> Mobile Voice Access Number