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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. When a device is roaming to a different site How does it get the Roaming Sensitive Settings from the Device Pool?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

Single Number Reach (Mobile Connect)

2. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?

To enable two-stage dialing.
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

3. What does an H323 GW require that MGCP GWs do not?
#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)#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
In the CUCM 7.0.1 Release Notes ... search for restore.
Dial-peers

4. What does Display-IE do?
#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
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
Sends the Calling Name.
Bottom up - to prevent glare.

5. How do you prevent toll fraud on CUCM?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
CUCM Service Parameters --> Call Park Reversion Timer
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True

6. How do you see multicast packets being sent?
Debug ephone moh
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
#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
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

7. How do you set the inter-digit timeout for SIP phones in CME?
(config)#voice register dialplan
Higher
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Master/Slave relationship. CUCM controls it.

8. 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
...
#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
IP Voice Media Streaming App
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

9. On an MGCP GW - if you select Top Down under Channel Selection Order - What does that mean?
Call forwarding between voip to voip (when CUBE is in play)
Top Down means that channel 1 will be the first channel used to place outgoing calls.
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
Bandwidth

10. What is a best practice for digit manipulation - in regards to H323 GWs?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Mobile Connect
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Re-enter the mgcp bind control and mgcp bind media commands.

11. What is Mobile Voice Access?
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.
(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
Debug isdn q931
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

12. What would force you to use telephony-service to configure SRST?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Max-dn|max-ephone|ip source-address
Device Pool Locations
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

13. What dtmf-relay type do you use for an H323 GW?
#test voice translation rule 1 <input to test>
On CUCM it's identical to adding an H323 GW.
Dtmf-relay h245-alpha
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.

14. On an H323 GW - how do you adjust the timers for redundancy hunting?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
'Service Parameters --> Auto Answer Timer
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3

15. How do you configure class of service (CoS) in CUCM? CME?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
1 ... this is not optional!
The name configuration field in ephone-dn and voice register 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

16. What are the steps to configure Single Number Reach and Mobile Voice Access?
Bandwidth
Assign SIP Dial Rules
At the CLI: utils ntp status

17. What are the commands to configure NTP in IOS?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Dtmf-relay h245-alpha
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

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

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19. How do you block an external call from being transferred back out to the pstn by an internal user?
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
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.
Service Parameter --> Block OffNet To OffNet Transfer --> True
To enable two-stage dialing.

20. How do you see the details of calls coming in and out of the PRI?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Debug isdn q931
Outbound dial-peers

21. What is the full E164 format?
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 ....
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

22. What are the commands to create vlans on an ESW?
1 ... this is not optional!
Put them in a route list.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
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 ....

23. An MGCP gateway serving as an SRST router requires what?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Configure it on the route group through the route list - then it will be local to the route list.
Re-enter the mgcp bind control and mgcp bind media commands.

24. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
8kb/s
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 forwarding between voip to voip (when CUBE is in play)

25. What CUCM services should you activate?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Device Pool Locations
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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

26. What are the steps to integrate CUPS with CUCM?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
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
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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)

27. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Re-enter the mgcp bind control and mgcp bind media commands.
Mobile Connect
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

28. How would you verify that DHCP is working in IOS?
...
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Mobile Voice Access
(config)#sh cdp neigh detail

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

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30. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
When configuring TEHO.
Matches any length dialed number and truncates it to 4 digits.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
(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

31. Two useful troubleshooting commands for CUCME?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
sh ephone reg|and|debug voip dialpeer
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

32. How would you enable security on a GK?
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
#voice service voip| #no supplementary-service h225-notify cid-update

33. With the gw-priority command - does higher or lower priority take precedence?
Outbound dial-peers
Higher
To enable two-stage dialing.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer

34. What are the 3 mandatory commands within call-manager-fallback?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Outbound dial-peers
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
Max-dn|max-ephone|ip source-address

35. How do you use an ephone template?
On CUCM it's identical to adding an H323 GW.
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.
Configure it on the route group through the route list - then it will be local to the route list.
Apply it to the ephone.

36. How much bandwidth does a G.729 call including layer 3 require?
24kb/s
Master/Slave relationship. CUCM controls it.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
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.

37. How do you enable AAR?
24kb/s
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Via the voice hunt-group parallel command

38. On the RDP What is the Rerouting CSS used for?
...
Single Number Reach (Mobile Connect)
Re-enter the mgcp bind control and mgcp bind media commands.
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.

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

(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
Max-dn|max-ephone|ip source-address
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

40. 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?
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.
When configuring TEHO.
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.
Bandwidth

41. What does ccm-manager config server [IP]</ip> do?
#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
Automatically configures the MGCP GW for you.
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
sh ephone reg|and|debug voip dialpeer

42. How do you support multiple codecs on a dialpeer?
(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
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 .)
...
Apply voice-class codec

43. What is the bit rate for a G.729 call excluding layer 2?
To notify SIP Phones of NTP
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
8kb/s
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

44. How do you configure phone ports on an ESW?
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
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Apply it to the ephone.
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

45. 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?
CUCM Service Parameters --> Call Park Reversion Timer
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
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 .)

46. Name 4 useful show commands for active calls.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
voice register pool|and|voice register dn
Bandwidth
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

47. What are the commands to configure an H323 GW?
Sh 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.
Assign SIP Dial Rules
#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

48. What's the difference between AAR and SRST?
(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
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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 ....
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

49. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Ccm-manager music-on-hold
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

50. How do you block calls under call-manager-fallback?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
#after-hours block pattern
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
To enable two-stage dialing.