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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 is the Service Parameter to allow the phone that is picking ip to auto answer?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
#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
Auto Call Pickup Enabled
Master/Slave relationship. CUCM controls it.

2. How do you use the # as a string terminator within a SIP Dial Rule?
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
#voice service voip| #no supplementary-service h225-notify cid-update
Precede the # with a > ... so 9011*>#

3. 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.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
CUCM OS Administration Settings --> NTP Servers
#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

4. How do you configure CUBE?

5. What does Display-IE do?
Sends the Calling Name.
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.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Auto Call Pickup Enabled

6. How do you support multiple codecs on a dialpeer?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Higher
Apply voice-class codec
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

7. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
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

Bottom up - to prevent glare.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1

8. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?
Dtmf-relay h245-alpha
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
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
Show ccm-manager

9. On the RDP What is the Rerouting CSS used for?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Call forwarding between voip to voip (when CUBE is in play)
When they are explicitly matched in a destination-pattern in a dial-peer.
Single Number Reach (Mobile Connect)

10. 'If you want to make changes to any softkeys where do you do it?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.

11. 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)
Call forwarding between voip to voip (when CUBE is in play)
#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
Call Simulator. You can use to validate path from router to the PSTN.

12. What does ccm-manager config server [IP]</ip> do?
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
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.
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
Automatically configures the MGCP GW for you.

13. What are the commands to configure a SIP phone in CUCME?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
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
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 ....
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

14. How do you get around relying on DNS for your CUCMs?
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 ....
Bottom up - to prevent glare.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

15. Two useful troubleshooting commands for CUCME?
sh ephone reg|and|debug voip dialpeer
To enable two-stage dialing.
(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
(config)#voice register dialplan

16. How much bandwidth does a G.729 call including layer 3 require?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
24kb/s

17. When do you use translate called? Translate calling?
24kb/s
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
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 .)

18. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
(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
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Dtmf-relay h245-alpha
The Called/Calling Transformations are superceded.

19. What terminology translates to SRST?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Bandwidth
Reroute when there is a WAN Outage.
#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

20. What are the commands to configure a T1/E1 PRI?
#after-hours block pattern
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
(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
Show ccm-manager

21. In CME Where is the Calling Name derived from?
The name configuration field in ephone-dn and voice register dn
1 ... this is not optional!
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
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.

22. 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
Allows you to transfer by only pressing the Transfer button once.
SIP Route Pattern over a SIP Trunk.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

23. What is CSIM?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Call Simulator. You can use to validate path from router to the PSTN.
'Use before the ... so XXXX
Bottom up - to prevent glare.

24. How do you prevent H323 caller-id updates to CUCM
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
#voice service voip| #no supplementary-service h225-notify cid-update
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au

25. Describe how you configure SIP URI functionality.
Assign SIP Dial Rules

#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
Outbound dial-peers

26. How can you confirm the MGCP GW is registered to CUCM - in IOS?
SIP Route Pattern over a SIP Trunk.
(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
Sh ccm-manager

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

28. With the gw-priority command - does higher or lower priority take precedence?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Higher
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Redundant dial plan. One on CUCM and one on the routers via dial-peers.

29. When using GK redundancy - what must you do to ensure both SUB and PUB use port 1720 for call-signalling.

30. What's the difference between AAR and SRST?
Service Parameter --> Block OffNet To OffNet Transfer --> True
Mobile Voice Access
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

31. How is NTP sych setup in CUCM?
Matches any length dialed number and truncates it to 4 digits.
CUCM OS Administration Settings --> NTP Servers
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
#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

32. Does CUCM support RSVP natively?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
CUCM Service Parameters --> Call Park Reversion Timer
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

33. How do you place SCCP and SIP phones into a single huntgroup?
Via the voice hunt-group parallel command
Dtmf-relay h245-alpha
(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
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

34. What are the basic ephone-dn and ephone commands?
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Sh ip rsvp reservation||sh sccp connections

35. 'How do you inform a SIP phone of NTP information?
TRUE
(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
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

36. How do you test a Voice Translation Rule?
Master/Slave relationship. CUCM controls it.
#test voice translation rule 1 <input to test>
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

37. How do you configure phone ports on a 3750?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

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
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast

38. What are the commands to configure a MGCP Gateway? (router)
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
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 .)
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
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.

39. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Configure it on the route group through the route list - then it will be local to the route list.
Ccm-manager music-on-hold
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
(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

40. Calls coming from CUCM to PSTN need what?
Master/Slave relationship. CUCM controls it.
Outbound dial-peers
Put them in a route list.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

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

42. What are the 3 mandatory commands within call-manager-fallback?
Max-dn|max-ephone|ip source-address
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
(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

43. Is the order of the MRGs in an MRGL significant?
Call forwarding between voip to voip (when CUBE is in play)
Master/Slave relationship. CUCM controls it.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.

44. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
To notify SIP Phones of NTP
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)#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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

45. 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
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
TRUE

46. How do you see the details of calls coming in and out of the PRI?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Debug isdn q931
sh ephone reg|and|debug voip dialpeer
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

47. How do you configure an MGCP GW? (router side)
In the CUCM 7.0.1 Release Notes ... search for restore.
(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 Dial Rules
#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

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

49. How do you verify that NTP is working on the CUCM server?
...
At the CLI: utils ntp status
In the CUCM 7.0.1 Release Notes ... search for restore.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True

50. What is Mobile Voice Access?
CUCM Service Parameters --> 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.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
When they are explicitly matched in a destination-pattern in a dial-peer.