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

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2. What terminology translates to SRST?
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
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
Reroute when there is a WAN Outage.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'

3. How do you enable Extension Mobility for a device in CME?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
1 ... this is not optional!
Assign a logout-profile to the ephone.

4. How do you disable KPML?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
(config)#sh cdp neigh detail
Assign SIP Dial Rules

5. When using drop-through-option What is the max number of huntgroups?
Call-manager-fallback or telephony-service mode SRST
When multiple sites use the same route pattern - do your digit manipulation on a route list.

1 ... this is not optional!

6. What does the Transfer On-hook Enabled Service Parameter do?
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
'Use before the ... so XXXX
Configure it on the route group through the route list - then it will be local to the route list.
Allows you to transfer by only pressing the Transfer button once.

7. What do you need to do to activate the CME GUI?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
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
TRUE
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

8. How would you enable security on a GK?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
At the CLI: utils ntp status
To enable two-stage dialing.
(config)#sh cdp neigh detail

9. How do you use an ephone template?
IP Voice Media Streaming App
Apply it to the ephone.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

10. Name 4 useful show commands for active calls.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Assign a logout-profile to the ephone.
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

11. 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
Single Number Reach (Mobile Connect)
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

12. How do you see the details of calls coming in and out of the PRI?
Outbound dial-peers
Debug isdn q931
Apply voice-class codec
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie

13. How do you allow H323 calls to be preserved should the primary H323 GW fail?
Assign a logout-profile to the ephone.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
The name configuration field in ephone-dn and voice register dn
The Called/Calling Transformations are superceded.

14. An MGCP gateway serving as an SRST router requires what?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Assign a logout-profile to the ephone.
Re-enter the mgcp bind control and mgcp bind media commands.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

15. When do you use translate called? Translate calling?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
(config)#voice register dialplan

16. 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
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
#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

17. What is a best practice for digit manipulation - in regards to H323 GWs?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.

18. In CME Where is the Calling Name derived from?
Automatically configures the MGCP GW for you.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
The name configuration field in ephone-dn and voice register dn
To enable two-stage dialing.

19. What is the bit rate for a G.729 call excluding layer 2?
In CUCM - configure CFUR to point to it's E164 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
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
8kb/s

20. 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
8kb/s
To enable two-stage dialing.
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

21. 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.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Put them in a route list.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.

22. How do you prevent H323 caller-id updates to CUCM
CUCM Service Parameters --> Call Park Reversion Timer
#voice service voip| #no supplementary-service h225-notify cid-update
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Top Down means that channel 1 will be the first channel used to place outgoing calls.

23. 'How do you inform a SIP phone of NTP information?
Apply voice-class codec
'Use before the ... so XXXX
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
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 .)

24. What are the commands to configure 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
Put them in a route list.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
#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

25. What does an H323 GW require that MGCP GWs do not?
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)
Dial-peers
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
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.

26. What is the Service Parameter to allow the phone that is picking ip to auto answer?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
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.
Auto Call Pickup Enabled

27. What does ccm-manager config server [IP]</ip> do?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Dtmf-relay h245-alpha
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Automatically configures the MGCP GW for you.

28. What's the difference between AAR and SRST?
Apply voice-class codec
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
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 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

29. 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
Sends the Calling Name.
...
The Called/Calling Transformations are superceded.
Via the voice hunt-group parallel command

30. What are the commands to create vlans on an ESW?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
SIP Route Pattern over a SIP Trunk.
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
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

31. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
Ccm-manager music-on-hold
SIP Route Pattern over a SIP Trunk.

Enable MOH on the CME.||#telephony service|#moh music-on-hold.au

32. What are two caveats to using the ccm-manager config server</ip> command?

Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
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 .)
Auto Call Pickup Enabled

33. When do you do digit manipulation at the route pattern as opposed to the route list?
Assign SIP Dial Rules
'Use before the ... so XXXX
Phone number followed by domain name.|i.e. 3006@ipxcme.com
When multiple sites use the same route pattern - do your digit manipulation on a route list.

34. When dealing with an MGCP GW there is no intelligence on the router. How would you strip an incoming Called Number from 10 to 4 digits?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
(config)#voice register dialplan
voice translation-profile|and|voice translation-rule

35. When a CUCM device dials a number - what happens?
Call Simulator. You can use to validate path from router to the PSTN.
Debug isdn q931
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 .)
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

36. What is SIP URI?
Apply voice-class codec
Phone number followed by domain name.|i.e. 3006@ipxcme.com
...
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones

37. How do you allow the Calling Name to be sent to the PSTN on a router?
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
Apply voice-class codec
#after-hours block pattern

38. 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?
Bandwidth
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
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)
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

39. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
Max-dn|max-ephone|ip source-address
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
(config)#sh cdp neigh detail
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323

40. How do you place SCCP and SIP phones into a single huntgroup?
Via the voice hunt-group parallel command
Reroute when here is WAN congestion.
On CUCM it's identical to adding an H323 GW.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files

41. How do you enable AAR system wide?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
SIP Route Pattern over a SIP Trunk.
In CUCM - configure CFUR to point to it's E164 number.

42. In Gatekeeper CAC how do you restrict a specific endpoint?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
The Called/Calling Transformations are superceded.
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

43. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
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
In CUCM - configure CFUR to point to it's E164 number.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Single Number Reach (Mobile Connect)

44. What CUCM services should you activate?
#after-hours block pattern
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
...
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

45. What is CSIM?
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.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
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

46. How do you set up redundancy on outbound dial-peers on an H323 gateway?
#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
#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
The name configuration field in ephone-dn and voice register dn

47. How do you configure an MGCP GW? (router side)
'Service Parameters --> Auto Answer Timer
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
(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
The Called/Calling Transformations are superceded.

48. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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.
#test voice translation rule 1 <input to test>

49. Maximum Wait Time for Desk Pickup?
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.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
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

50. 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
TRUE
Reroute when here is WAN congestion.