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. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
The Called/Calling Transformations are superceded.
(config)#voice register dialplan
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow


2. In Gatekeeper CAC how do you restrict a specific endpoint?
Sends the Calling Name.
(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!
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones


3. What is CSIM?
Call Simulator. You can use to validate path from router to the PSTN.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
When multiple sites use the same route pattern - do your digit manipulation on a route list.


4. What are the commands to configure a T1/E1 PRI?
(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
Matches any length dialed number and truncates it to 4 digits.
Create 2 voip dial-peers - one the Sub and one to the Pub (the Pub with a preference of 1). Then create a <i>voice class h323</i>and set the <i>h225 timeout</i> value so it will hunt between the two.
Dtmf-relay h245-alpha


5. What is SIP URI?
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.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
The Called/Calling Transformations are superceded.
Dtmf-relay h245-alpha


6. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?


7. How can you confirm the MGCP GW is registered to CUCM - in IOS?
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Sh ccm-manager
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
<i>SIP Route Pattern</i> over a SIP Trunk.


8. 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
#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
Phone number followed by domain name.|i.e. 3006@ipxcme.com
(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


9. When a CUCM device dials a number - what happens?

It's best to strip digits at the voice port.
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 .)
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>


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


11. Name 4 useful show commands for active calls.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
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.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary


12. When using <i>drop-through-option</i> What is the max number of huntgroups?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Call Simulator. You can use to validate path from router to the PSTN.
TRUE
1 ... this is not optional!


13. How much bandwidth does a G.729 call including layer 3 require?
<i>voice register pool</i>|and|<i>voice register dn</i>
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
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
24kb/s


14. How do you configure class of service (CoS) in CUCM? CME?
(config)#voice register dialplan
Call Simulator. You can use to validate path from router to the PSTN.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists


15. How would you enable security on a GK?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
IP Voice Media Streaming App
(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
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage


16. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
(config)#<i>sh cdp neigh detail
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
It's best to strip digits at the voice port.
Ccm-manager music-on-hold


17. What does an H323 GW require that MGCP GWs do not?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Dial-peers
Debug isdn q931
Apply <i>voice-class codec</i>


18. How do you see multicast packets being sent?
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
Sends the Calling Name.
Debug ephone moh
Assign SIP Dial Rules


19. How do you block an external call from being transferred back out to the pstn by an internal user?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Reroute when there is a WAN Outage.
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
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


20. How do you configure phone ports on an ESW?
Call forwarding between voip to voip (when CUBE is in play)
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
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
The Called/Calling Transformations are superceded.


21. How do you prevent toll fraud on CUCM?
...
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True


22. What are the commands to configure a SIP phone in CUCME?
(config)#voice register dialplan
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
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST


23. How do you disable KPML?
Debug ephone moh
Assign SIP Dial Rules
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
<i>SIP Route Pattern</i> over a SIP Trunk.


24. What are the commands to create vlans on an ESW?
#voice service voip| #no supplementary-service h225-notify cid-update
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
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>


25. How do you see the details of calls coming in and out of the 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
TRUE
#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
Debug isdn q931


26. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
Create 2 voip dial-peers - one the Sub and one to the Pub (the Pub with a preference of 1). Then create a <i>voice class h323</i>and set the <i>h225 timeout</i> value so it will hunt between the two.
<i>voice register pool</i>|and|<i>voice register dn</i>
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Debug isdn q931


27. 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?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
CUCM OS Administration Settings --> NTP Servers
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group


28. What are the steps to integrate CUPS with CUCM?
On CUCM -- UPTSAL||Create Your <b>Users</b>|<b>Phone</b> - set owner id|Create SIP <b>Trunk</b>|<b>SIP Trunk Security Profile|</b>|<b>Application Server</b>|<b>Licensing|</b>|||On CUPS||Work Left to right
Dial-peers
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
1 ... this is not optional!


29. How do you set up redundancy on outbound dial-peers on an H323 gateway?
<i>voice register pool</i>|and|<i>voice register dn</i>
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Higher


30. In CME Where is the Calling Name derived from?
'Service Parameters --> <i>Auto Answer Timer</i>
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
The <i>name</i> configuration field in ephone-dn and voice register dn
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>


31. 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.
Precede the # with a > ... so 9011*>#
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
8kb/s


32. What are the commands to configure a MGCP Gateway? (router)
Via the <i>voice hunt-group parallel</i> command
'Use before the ... so XXXX
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
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


33. How do you configure AAR?
Assign a logout-profile to the ephone.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
The <i>name</i> configuration field in ephone-dn and voice register dn
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.


34. What are the basic ephone-dn and ephone commands?
Master/Slave relationship. CUCM controls it.
On CUCM -- UPTSAL||Create Your <b>Users</b>|<b>Phone</b> - set owner id|Create SIP <b>Trunk</b>|<b>SIP Trunk Security Profile|</b>|<b>Application Server</b>|<b>Licensing|</b>|||On CUPS||Work Left to right
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
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.


35. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
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
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM


36. How do you block calls under call-manager-fallback?
At the CLI: <i>utils ntp status</i>
<i>#after-hours block pattern</i>
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Sh ip rsvp reservation||sh sccp connections


37. <b>By Nov. 16 - 2010</b>||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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
#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
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
...


38. What is Phone NTP Reference used for?
Sh ip rsvp reservation||sh sccp connections
Sh ccm-manager
To notify SIP Phones of NTP
(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


39. What would force you to use telephony-service to configure SRST?
Master/Slave relationship. CUCM controls it.
Mobile Connect
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
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup


40. When do you do digit manipulation at the route pattern as opposed to the route list?
'Use before the ... so XXXX
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 multiple sites use the same route pattern - do your digit manipulation on a route list.
Reroute when here is WAN congestion.


41. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
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
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Service Parameters --> <i>Mobile Voice Access Number</i>
Top Down means that channel 1 will be the first channel used to place outgoing calls.


42. 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
Master/Slave relationship. CUCM controls it.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3


43. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
To enable two-stage dialing.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
#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


44. How do you configure CUCM redundancy on an H323 gateway?
Single Number Reach (Mobile Connect)
Phone number followed by domain name.|i.e. 3006@ipxcme.com
#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
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.


45. How is NTP sych setup in CUCM?
CUCM OS Administration Settings --> NTP Servers
#voice service voip| #no supplementary-service h225-notify cid-update
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Apply <i>voice-class codec</i>


46. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
It's best to strip digits at the voice port.
#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
Bottom up - to prevent glare.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify


47. What are the commands to manually configure an MGCP gateway?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
#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

8kb/s


48. On the RDP What is the CSS used for?
#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
Mobile Voice Access
Reroute when here is WAN congestion.
Put them in a route list.


49. How do you configure a SIP Trunk? (router side)
(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
TRUE
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True


50. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
<i>#after-hours block pattern</i>
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
(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
When they are explicitly matched in a destination-pattern in a dial-peer.