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CCIE Voice Test

Instructions:

Answer 50 questions in 15 minutes.
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Match each statement with the correct term.
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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 are the commands to manually configure an MGCP gateway?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
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
Sh ip rsvp reservation||sh sccp connections
#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

2. What terminology translates to AAR?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Reroute when here is WAN congestion.
Sh ccm-manager
Debug isdn q931

3. When configuring a Gatekeep-controlled trunk - what do you do with the Wait for Far End H.245 TCS option?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Sh ip rsvp reservation||sh sccp connections

4. What are the commands to set up the PRI?

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5. 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?
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)
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
#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

6. 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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
At the CLI: utils ntp status

7. What terminology translates to SRST?
voice translation-profile|and|voice translation-rule
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
'Service Parameters --> Auto Answer Timer
Reroute when there is a WAN Outage.

8. What CM Service needs to be start in Serviceability for MOH to work?
...
IP Voice Media Streaming App
Debug isdn q931
8kb/s

9. SNR is also known as?
Ccm-manager music-on-hold
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
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Mobile Connect

10. How do do configure TEHO?
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
24kb/s
Service Parameter --> Block OffNet To OffNet Transfer --> 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.

11. 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
#test voice translation rule 1 <input to test>
Call-manager-fallback or telephony-service mode SRST
Debug ephone moh

12. How do you configure AAR?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
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.
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

13. How would you enable security on a GK?

1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
On CUCM it's identical to adding an H323 GW.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow

14. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
SIP Dial Rules
Device Pool Locations
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Single Number Reach (Mobile Connect)

15. 'If you're not getting a DHCP address from CUCM what then?
#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)#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
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|utils csa disable
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.

16. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
Sends the Calling Name.
Mobile Voice Access
SIP Route Pattern over a SIP Trunk.
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

17. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
...
When configuring TEHO.

18. When are digits stripped in a gateway?
Master/Slave relationship. CUCM controls it.
CUCM Service Parameters --> Automated Alternate Routing Enable --> True
When they are explicitly matched in a destination-pattern in a dial-peer.
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

19. What is the Enable Enterprise Feature Access Service Parameter used for?

CUCM Service Parameters --> Call Park Reversion Timer
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
To enable two-stage dialing.

20. Is the order of the MRGs in an MRGL significant?
CUCM Service Parameters --> Call Park Reversion Timer
Automatically configures the MGCP GW for you.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Call-manager-fallback or telephony-service mode SRST

21. What is SIP URI?
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
The name configuration field in ephone-dn and voice register dn
Phone number followed by domain name.|i.e. 3006@ipxcme.com
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.

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 os truly unsurpassed. I have opened up a wealth of opportunities for myself and my family and
...
Call forwarding between voip to voip (when CUBE is in play)
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
'Service Parameters --> Auto Answer Timer

23. What does KPML do?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
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.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Assign a logout-profile to the ephone.

24. How can you confirm the MGCP GW is registered to CUCM - in IOS?
It's best to strip digits at the voice port.
Call-manager-fallback or telephony-service mode SRST
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Sh ccm-manager

25. When using drop-through-option What is the max number of huntgroups?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Bottom up - to prevent glare.
Dtmf-relay h245-alpha
1 ... this is not optional!

26. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
Show ccm-manager
Mobile Voice Access
voice register pool|and|voice register dn

27. How do you create a trunk on the switch side?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!

28. What types of digit manipulation can you perform at the route pattern?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
(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
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
#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

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

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30. What are the commands to configure a SIP phone in CUCME?
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
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Higher
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

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

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32. What are the steps to integrate CUPS with CUCM?
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
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.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)

33. With the gw-priority command - does higher or lower priority take precedence?
In CUCM - configure CFUR to point to it's E164 number.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Higher

34. 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
#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
CAT IOS|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
(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

35. 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.
Dial-peers
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
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

36. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
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
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
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

37. How do you configure phone ports on an ESW?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Enable MOH on the CME.||#telephony service|#moh music-on-hold.au
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

38. How do you prioritize route groups?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
#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
Call-manager-fallback or telephony-service mode SRST
Put them in a route list.

39. Does CUCM support RSVP natively?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Single Number Reach (Mobile Connect)
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
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 ....

40. A show isdn status shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Higher
Re-enter the mgcp bind control and mgcp bind media commands.
SIP Route Pattern over a SIP Trunk.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False

41. What are the commands to create the L3 routing interface for VLANS (SVI)?
Mobile Connect
voice register pool|and|voice register dn
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
Debug isdn q931

42. What's the difference between AAR and SRST?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
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
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 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

43. Where do you use VIA zone?
Reroute when there is a WAN Outage.
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
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
(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

44. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Bottom up - to prevent glare.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
System Cisco Unified CM|Uncheck the box Auto-Reg Disabled||Enterprise Parameters --> Auto Registration Phone Protocol SIP or SCCP

45. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
Default is SRST||mode cme changes to CME|no mode cme changes back to SRST
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
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 the CUCM 7.0.1 Release Notes ... search for restore.

46. How do you verify that NTP is working on the CUCM server?
Ccm-manager music-on-hold
At the CLI: utils ntp status
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
8kb/s

47. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
Proxy Line Mode has multiple icons on the phone.|Shared Line Mode has only one.
Call Simulator. You can use to validate path from router to the PSTN.
The name configuration field in ephone-dn and voice register dn
Put them in a route list.

48. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
#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

49. What is the full E164 format?
24kb/s
Outbound dial-peers
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
(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

50. On the RDP What is the CSS used for?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Dial-peers
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

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