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CCIE Voice Test

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Answer 50 questions in 15 minutes.
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1. What are the commands to configure a SIP phone in CUCME?
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
Sh ip rsvp reservation||sh sccp connections
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
CUCM OS Administration Settings --> NTP Servers

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

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3. What does KPML do?
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
(config)#sh cdp neigh detail
Mobile Voice Access
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.

4. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
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
CUCM OS Administration Settings --> NTP Servers
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
Mobile Connect

5. What would force you to use telephony-service to configure SRST?
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
'Use before the ... so XXXX
Dial-peers
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup

6. Describe the relationship between route patterns and end devices.
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Apply voice-class codec
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.

7. 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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Higher
Debug ephone moh

8. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
The Called/Calling Transformations are superceded.
8kb/s
The name configuration field in ephone-dn and voice register dn
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

9. In CME Where is the Calling Name derived from?
Master/Slave relationship. CUCM controls it.
Single Number Reach (Mobile Connect)
Matches any length dialed number and truncates it to 4 digits.
The name configuration field in ephone-dn and voice register dn

10. 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
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.
Ccm-manager music-on-hold

11. When a device is roaming to a different site How does it get the Roaming Sensitive Settings from the Device Pool?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Allows you to transfer by only pressing the Transfer button once.
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

12. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
SIP Route Pattern over a SIP Trunk.
voice register pool|and|voice register dn
Access Port Method||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
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

13. When setting up SIP URI where do you configure the CUCM's domain name?'
(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
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Dial-peers

14. How do you prioritize route groups?
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
When configuring TEHO.
Put them in a route list.
When they are explicitly matched in a destination-pattern in a dial-peer.

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 os truly unsurpassed. I have opened up a wealth of opportunities for myself and my family and
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
...
Max-dn|max-ephone|ip source-address
(config)#sh cdp neigh detail

16. '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
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.
...
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Configure it on the route group through the route list - then it will be local to the route list.

17. 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
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
#test voice translation rule 1 <input to test>
Allows you to transfer by only pressing the Transfer button once.

18. 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.
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)
voice register pool|and|voice register dn
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs

19. How do you allow H323 calls to be preserved should the primary H323 GW fail?
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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

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

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21. SNR is also known as?
Mobile Connect
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
SIP Route Pattern over a SIP Trunk.
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

22. What are the commands to configure a T1/E1 PRI?
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
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.
(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
Redundant dial plan. One on CUCM and one on the routers via dial-peers.

23. How do you use an ephone template?
#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
Call-manager-fallback or telephony-service mode SRST
Apply it to the ephone.
Show ccm-manager

24. Name 4 useful show commands for active calls.
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 ....
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Sh ip rsvp reservation||sh sccp connections
Configure it on the route group through the route list - then it will be local to the route list.

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

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

(config)#sh cdp neigh detail
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 ....
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).

27. What are two caveats to using the ccm-manager config server</ip> command?
...
On CUCM it's identical to adding an H323 GW.

Sh ccm-manager

28. How do you configure CUBE?

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29. 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?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.

30. How do you setup AutoRegistration in CUCM?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
8kb/s
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

31. What commands are needed to configure the voice register pool in CME?
#test voice translation rule 1 <input to test>
(config)#sh cdp neigh detail
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
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

32. How do you enable AAR system wide?
To enable two-stage dialing.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
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 .)
At the CLI: utils ntp status

33. When do you do digit manipulation at the route pattern as opposed to the route list?
Apply voice-class codec
When multiple sites use the same route pattern - do your digit manipulation on a route list.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

34. How do you place SCCP and SIP phones into a single huntgroup?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Via the voice hunt-group parallel command

No POTS dial-peer or no direct-inward-dial on a pots dial-peer

35. How do you configure CUCM redundancy on an H323 gateway?
#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
'Use before the ... so XXXX
Debug isdn q931
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)

36. How do you see the details of calls coming in and out of the PRI?
Debug isdn q931
Device Pool Locations
...
sh ephone reg|and|debug voip dialpeer

37. How do you verify that NTP is working on the CUCM server?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Enterprise Parameters --> Organization Top Level Domain|Enterprise Parameters --> Cluster Fully Qualified Domain Name
At the CLI: utils ntp status
CUCM Service Parameters --> Call Park Reversion Timer

38. When a CUCM device dials a number - what happens?
Assign a logout-profile to the ephone.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
To notify SIP Phones of NTP
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 .)

39. 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?
To enable two-stage dialing.
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)
#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
Debug isdn q931

40. On an H323 GW - how do you adjust the timers for redundancy hunting?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Dial-peers
h225 timeout tcp establish 3 within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
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

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

42. On an MGCP GW - how could you see the primary and backup CUCM servers?
Ccm-manager music-on-hold
Show ccm-manager
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

43. What are the 3 mandatory commands within call-manager-fallback?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Max-dn|max-ephone|ip source-address
CUCM Service Parameters --> Call Park Reversion Timer
...

44. 'If you're not getting a DHCP address from CUCM what then?
Sh ip rsvp reservation||sh sccp connections
Service Parameters --> Mobile Voice Access Number
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable

45. Name 2 commands to verify RSVP functionality.
#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
Ccm-manager music-on-hold
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Sh ip rsvp reservation||sh sccp connections

46. What CUCM services should you activate?
At the CLI: utils ntp status
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Call-manager-fallback or telephony-service mode SRST

47. What are the commands to create vlans on an ESW?

Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Max-dn|max-ephone|ip source-address

48. What CM Service needs to be start in Serviceability for MOH to work?
Allows you to transfer by only pressing the Transfer button once.
IP Voice Media Streaming App
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
(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

49. When can you not use a Standard Local Route Group?
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
24kb/s
Bandwidth
When configuring TEHO.

50. What should you configure before entering auto qos?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Debug ephone moh
Dial-peers
Bandwidth

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CCIE Voice Test

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