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Test your basic knowledge |
CCIE Voice Test
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Study First
Subjects
:
cisco
,
it-skills
,
ccie
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 does <i>rule 1 /^.*(....$)/ /1/</i> do?
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.
Apply <i>voice-class codec</i>
In CUCM - configure CFUR to point to it's E164 number.
Matches any length dialed number and truncates it to 4 digits.
2. What does an H323 GW require that MGCP GWs do not?
Dial-peers
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 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
It's best to strip digits at the voice port.
3. How would you set the the timer for Auto Answer?
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4. How do you configure CUBE?
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5. What should you configure before entering auto qos?
Bandwidth
(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
Dial-peers
Matches any length dialed number and truncates it to 4 digits.
6. How do you see the details of calls coming in and out of the PRI?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Debug isdn q931
#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
<i>voice register pool</i>|and|<i>voice register dn</i>
7. 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?
Single Number Reach (Mobile Connect)
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
8. How do you configure phone ports on an ESW?
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
Apply <i>voice-class codec</i>
Master/Slave relationship. CUCM controls it.
Reroute when here is WAN congestion.
9. How do you setup AutoRegistration in CUCM?
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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
10. Is the order of the MRGs in an MRGL significant?
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
(config)#voice register dialplan
The <i>name</i> configuration field in ephone-dn and voice register dn
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
11. What terminology translates to AAR?
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
Apply it to the ephone.
Reroute when here is WAN congestion.
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 ....
12. How do you place SCCP and SIP phones into a single huntgroup?
Via the <i>voice hunt-group parallel</i> command
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
It's best to strip digits at the voice port.
13. What kind of relationship does an MGCP gateway have with CUCM?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Dtmf-relay h245-alpha
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Master/Slave relationship. CUCM controls it.
14. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
Call Simulator. You can use to validate path from router to the PSTN.
...
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
15. What are the commands to create vlans on an ESW?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
SIP Dial Rules
16. What do you need to do to activate the CME GUI?
Debug ephone moh
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
Find the GUI Files - show flash | i .html|<i>ip http server</i>|<i>ip http path flash:/<PATH TO FILES></i>||telephony service|web admin system name admin pass cisco|dn-webedit
17. In CME Where is the Calling Name derived from?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Reroute when there is a WAN Outage.
TRUE
The <i>name</i> configuration field in ephone-dn and voice register dn
18. In Gatekeeper CAC how do you restrict a specific endpoint?
In CUCM - configure CFUR to point to it's E164 number.
Bandwidth
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.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
19. How much bandwidth does a G.729 call including layer 3 require?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Bottom up - to prevent glare.
24kb/s
20. How do you enable Extension Mobility for a device in CME?
CUCM OS Administration Settings --> NTP Servers
Allows you to transfer by only pressing the Transfer button once.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Assign a logout-profile to the ephone.
21. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Ccm-manager music-on-hold
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
22. What is a best practice for digit manipulation - in regards to H323 GWs?
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
23. What is the bit rate for a G.729 call excluding layer 2?
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.
1 ... this is not optional!
(config)#<i>sh cdp neigh detail
8kb/s
24. How do you create a trunk on the router side?
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
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
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
25. On the RDP What is the CSS used for?
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Mobile Voice Access
Call forwarding between voip to voip (when CUBE is in play)
SERVICE PARAMETER> Automated Alternate Routing Enable > True
26. What would force you to use telephony-service to configure SRST?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
#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
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
27. 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.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
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
Configure it on the route group through the route list - then it will be local to the route list.
28. What does the <i>Transfer On-hook Enabled</i> Service Parameter do?
(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
Allows you to transfer by only pressing the Transfer button once.
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
Apply <i>voice-class codec</i>
29. How do you use an ephone template?
(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
Outbound dial-peers
Apply it to the ephone.
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
30. SNR is also known as?
Master/Slave relationship. CUCM controls it.
Mobile Connect
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Service Parameters --> <i>Mobile Voice Access Number</i>
31. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
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
Top Down means that channel 1 will be the first channel used to place outgoing calls.
It's best to strip digits at the voice port.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
32. How do you set up redundancy on outbound dial-peers on an H323 gateway?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
...
<i>#after-hours block pattern</i>
33. How do you test a Voice Translation Rule?
Sh ccm-manager
#test voice translation rule 1 <input to test>
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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
34. How do you configure class of service (CoS) in CUCM? CME?
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Higher
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Matches any length dialed number and truncates it to 4 digits.
35. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
<i>voice register pool</i>|and|<i>voice register dn</i>
At the CLI: <i>utils ntp status</i>
Sh ccm-manager
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
36. What is call-forward pattern used for?
Via the <i>voice hunt-group parallel</i> command
Apply <i>voice-class codec</i>
Call forwarding between voip to voip (when CUBE is in play)
Dial-peers
37. Two useful troubleshooting commands for CUCME?
(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
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.
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
38. What are two caveats to using the <i>ccm-manager config server</ip> command?
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
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
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. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
Allows you to transfer by only pressing the Transfer button once.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
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
<i>SIP Route Pattern</i> over a SIP Trunk.
40. How do you configure an MGCP GW? (router side)
(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
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
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
24kb/s
41. How is NTP sych setup in CUCM?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
At the CLI: <i>utils ntp status</i>
Mobile Connect
CUCM OS Administration Settings --> NTP Servers
42. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
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43. 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.
When they are explicitly matched in a destination-pattern in a dial-peer.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
44. What are the base telephony-service commands for CME?
To enable two-stage dialing.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Apply <i>voice-class codec</i>
Reroute when here is WAN congestion.
45. When configuring a Gatekeep-controlled trunk - what do you do with the <i>Wait for Far End H.245 TCS</i> option?
Use the <i>alias static</i> configuration||alias static (IP of CUCM) gkid (GK ID) gateway voip ras (IP of CUCM) e164 (dn) e164 (dn) e164 (dn)
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Dial-peers
When multiple sites use the same route pattern - do your digit manipulation on a route list.
46. What CUCM services should you activate?
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 ....
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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
47. How do you configure a SIP Trunk? (router side)
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
(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
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
(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
48. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
1 ... this is not optional!
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
49. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
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 <i>restore</i>.
50. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
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.
To enable two-stage dialing.
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
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
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