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CCIE Voice Test

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Answer 50 questions in 15 minutes.
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Match each statement with the correct term.
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1. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?

#test voice translation rule 1 <input to test>
Higher
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.

2. How do you configure CUBE?

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3. How do you configure SRST?
Auto Call Pickup Enabled
The name configuration field in ephone-dn and voice register dn
At the CLI: utils ntp status
Call-manager-fallback or telephony-service mode SRST

4. How would you verify that DHCP is working in IOS?
SIP Dial Rules
#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
(config)#sh cdp neigh detail
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

5. What are two caveats to using the ccm-manager config server</ip> command?
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

To notify SIP Phones of NTP
Mobile Connect

6. How is NTP sych setup in CUCM?
CUCM OS Administration Settings --> NTP Servers
Dtmf-relay h245-alpha

...

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

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8. How do you configure an MGCP GW? (router side)
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
(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
Bandwidth

9. What is Mobile Voice Access?
Ccm-manager music-on-hold
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
To notify SIP Phones of NTP

10. How do you create a trunk on the switch side?
Master/Slave relationship. CUCM controls it.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

11. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
Auto Call Pickup Enabled
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
Assign a logout-profile to the ephone.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True

12. Describe how you configure SIP URI functionality.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

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.
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.

13. How would you enable security on a GK?
Device Pool Locations
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
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

14. How do you set the Call Park Reversion Timer?
Auto Call Pickup Enabled
Higher
CUCM Service Parameters --> Call Park Reversion Timer
Assign SIP Dial Rules

15. Two useful troubleshooting commands for CUCME?
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.
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.

sh ephone reg|and|debug voip dialpeer

16. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
In CUCM - configure CFUR to point to it's E164 number.
Max-dn|max-ephone|ip source-address
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

17. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
(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
In the CUCM 7.0.1 Release Notes ... search for restore.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
#test voice translation rule 1 <input to test>

18. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
'Service Parameters --> Auto Answer Timer
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
The Called/Calling Transformations are superceded.
Re-enter the mgcp bind control and mgcp bind media commands.

19. What do you need to do to activate the CME GUI?
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
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Higher
24kb/s

20. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
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
Bottom up - to prevent glare.
(config)#sh cdp neigh detail

21. When using drop-through-option What is the max number of huntgroups?
To notify SIP Phones of NTP
1 ... this is not optional!
(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
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

22. 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
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
...
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

23. How do you configure a gateway to register with gatekeeper?
#voice service voip| #no supplementary-service h225-notify cid-update
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
#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
Top Down means that channel 1 will be the first channel used to place outgoing calls.

24. What is CSIM?
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.
Call Simulator. You can use to validate path from router to the PSTN.
Dial-peers
Phone number followed by domain name.|i.e. 3006@ipxcme.com

25. Maximum Wait Time for Desk Pickup?
...
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Higher

26. 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?
#voice service voip| #no supplementary-service h225-notify cid-update

SIP Dial Rules
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.

27. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Precede the # with a > ... so 9011*>#
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
(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

28. How do you test a Voice Translation Rule?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
SIP Route Pattern over a SIP Trunk.
#test voice translation rule 1 <input to test>
Outbound dial-peers

29. When setting up SIP URI where do you configure the CUCM's domain name?'
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.

30. Where do you use VIA zone?
(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
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
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
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

31. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?
24kb/s
Dial-peers
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
In the CUCM 7.0.1 Release Notes ... search for restore.

32. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Assign a logout-profile to the ephone.
voice register pool|and|voice register dn
Ccm-manager music-on-hold
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.

33. How do you prevent toll fraud on CME?

h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
#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 ....

34. What is SIP URI?
Assign a logout-profile to the ephone.
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
...
Phone number followed by domain name.|i.e. 3006@ipxcme.com

35. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
Master/Slave relationship. CUCM controls it.
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.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

36. What is Phone NTP Reference used for?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
To notify SIP Phones of NTP
Call-manager-fallback or telephony-service mode SRST
TRUE

37. Name 4 useful show commands for active calls.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Master/Slave relationship. CUCM controls it.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary

38. How do you configure DHCP in IOS?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Sends the Calling Name.
(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

39. How do you support multiple codecs on a dialpeer?
Apply voice-class codec
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
24kb/s
IP Voice Media Streaming App

40. What does Display-IE do?
Auto Call Pickup Enabled
(config)#sh cdp neigh detail
Sends the Calling Name.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.

41. What are the basic ephone-dn and ephone commands?
Max-dn|max-ephone|ip source-address
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
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

42. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
voice register pool|and|voice register dn
When multiple sites use the same route pattern - do your digit manipulation on a route list.
To enable two-stage dialing.

43. What terminology translates to AAR?
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
Reroute when here is WAN congestion.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Reroute when there is a WAN Outage.

44. How do you get around relying on DNS for your CUCMs?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
CUCM OS Administration Settings --> NTP Servers
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

45. How do you verify that NTP is working on the CUCM server?
At the CLI: utils ntp status
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
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.
#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

46. In Gatekeeper CAC how do you restrict a specific endpoint?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
voice translation-profile|and|voice translation-rule

47. What is the Service Parameter to allow the phone that is picking ip to auto answer?
Auto Call Pickup Enabled
To enable two-stage dialing.
Dial-peers
It's best to strip digits at the voice port.

48. '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 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
Outbound dial-peers
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

49. How do you set the inter-digit timeout for SIP phones in CME?
CUCM Service Parameters --> 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.
(config)#voice register dialplan
Higher

50. How do you use the # as a string terminator within a SIP Dial Rule?
Ccm-manager music-on-hold
Precede the # with a > ... so 9011*>#
(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
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.

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CCIE Voice Test

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