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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 dtmf-relay type do you use for an H323 GW?
Call-manager-fallback or telephony-service mode SRST
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
24kb/s
Dtmf-relay h245-alpha

2. What are the commands to configure an H323 GW?

...
#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
CUCM Service Parameters --> Call Park Reversion Timer

3. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
#after-hours block pattern
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
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
It's best to strip digits at the voice port.

4. On an MGCP GW - how could you see the primary and backup CUCM servers?
Show ccm-manager
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Ccm-manager music-on-hold
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

5. Where can you assign the AAR Group?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
IP Voice Media Streaming App
#voice service voip| #no supplementary-service h225-notify cid-update
Call forwarding between voip to voip (when CUBE is in play)

6. How do you support multiple codecs on a dialpeer?
Apply voice-class codec
Outbound dial-peers
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.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable

7. What types of digit manipulation can you perform at the route pattern?
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Mobile Voice Access
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

8. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
voice register pool|and|voice register dn
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
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.

9. With the gw-priority command - does higher or lower priority take precedence?
Higher
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.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

10. How can you confirm the MGCP GW is registered to CUCM - in IOS?
Sh ccm-manager
When multiple sites use the same route pattern - do your digit manipulation on a route list.
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.
The name configuration field in ephone-dn and voice register dn

11. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
It's best to strip digits at the voice port.
#after-hours block pattern
SIP Route Pattern over a SIP Trunk.
Dtmf-relay h245-alpha

12. How do you configure CUBE?

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13. On an H323 GW - how do you adjust the timers for redundancy hunting?
(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
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Service Parameters --> Mobile Voice Access Number
Top Down means that channel 1 will be the first channel used to place outgoing calls.

14. What are the steps to integrate CUPS with CUCM?
Call-manager-fallback or telephony-service mode SRST
To notify SIP Phones of NTP
(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
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

15. 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
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Sends the Calling Name.
...
Apply voice-class codec

16. How do you block calls under call-manager-fallback?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
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.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
#after-hours block pattern

17. 'If you're not getting a DHCP address from CUCM what then?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Sends the Calling Name.
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

18. What are the commands to configure SRST in fallback?
(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
...
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
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 ....

19. How do you set the inter-digit timeout for SIP phones in CME?
(config)#voice register dialplan
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
#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
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST

20. How do you use an ephone template?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Apply it to the ephone.
(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

21. When using drop-through-option What is the max number of huntgroups?
1 ... this is not optional!
Top Down means that channel 1 will be the first channel used to place outgoing calls.
SIP Dial Rules
The name configuration field in ephone-dn and voice register dn

22. What is SIP URI?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.

Phone number followed by domain name.|i.e. 3006@ipxcme.com
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM

23. 'How do you inform a SIP phone of NTP information?
Apply it to the ephone.
Device Pool Locations
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 ....
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

24. 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?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Sh ccm-manager
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
(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

25. '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
...
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.
Apply voice-class codec

26. What should you configure before entering auto qos?
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
#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

Bandwidth

27. How do you prevent toll fraud on CUCM?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|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.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
24kb/s

28. How is NTP sych setup in CUCM?
CUCM OS Administration Settings --> NTP Servers
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Call Simulator. You can use to validate path from router to the PSTN.
Mobile Voice Access

29. What are the 3 mandatory commands within call-manager-fallback?
Max-dn|max-ephone|ip source-address
Reroute when there is a WAN Outage.
Debug ephone moh
1 ... this is not optional!

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

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31. In CME Where is the Calling Name derived from?
Configure it on the route group through the route list - then it will be local to the route list.
The name configuration field in ephone-dn and voice register dn
Matches any length dialed number and truncates it to 4 digits.
Allows you to transfer by only pressing the Transfer button once.

32. How do you test a Voice Translation Rule?
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
#test voice translation rule 1 <input to test>
In the CUCM 7.0.1 Release Notes ... search for restore.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

33. How do you block an external call from being transferred back out to the pstn by an internal user?
#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
Service Parameter --> Block OffNet To OffNet Transfer --> True
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
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. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

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.

35. What terminology translates to AAR?
Reroute when here is WAN congestion.
...
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.

36. Which takes precedence Device Locations or Device Pool Locations?
Device Pool Locations
Master/Slave relationship. CUCM controls it.
Service Parameters --> Mobile Voice Access Number
Precede the # with a > ... so 9011*>#

37. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
It's best to strip digits at the voice port.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.

38. What is the Enable Enterprise Feature Access Service Parameter used for?
Configure it on the route group through the route list - then it will be local to the route list.
Automatically configures the MGCP GW for you.
Show ccm-manager
To enable two-stage dialing.

39. How do you ensure that G.711 only is used?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec Disable
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.
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.

40. When a CUCM device dials a number - what happens?
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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
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 .)
To notify SIP Phones of NTP

41. 'If you want to make changes to any softkeys where do you do it?
Via the voice hunt-group parallel command
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.

42. How do you setup AutoRegistration in CUCM?
Max-dn|max-ephone|ip source-address
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP
8kb/s
When multiple sites use the same route pattern - do your digit manipulation on a route list.

43. What are the commands to create the L3 routing interface for VLANS (SVI)?
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group

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

Phone number followed by domain name.|i.e. 3006@ipxcme.com
(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

45. Name 4 useful show commands for active calls.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
#test voice translation rule 1 <input to test>
Via the voice hunt-group parallel command
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

46. How do you enable AAR system wide?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Outbound dial-peers
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
Mobile Voice Access

47. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
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
(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
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.
Re-enter the mgcp bind control and mgcp bind media commands.

48. How do you set up redundancy on outbound dial-peers on an H323 gateway?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
...

49. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
#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
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage

50. What are the steps to configure Single Number Reach and Mobile Voice Access?
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

Call Simulator. You can use to validate path from router to the PSTN.
When they are explicitly matched in a destination-pattern in a dial-peer.