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CCIE Voice Test
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cisco
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it-skills
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ccie
Instructions:
Answer 50 questions in 15 minutes.
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study here
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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. Name 4 useful show commands for active calls.
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
'Use before the ... so XXXX
Dial-peers
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
2. When setting up SIP URI where do you configure the CUCM's domain name?'
Higher
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
It's best to strip digits at the voice port.
3. SNR is also known as?
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
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
'Service Parameters --> <i>Auto Answer Timer</i>
Mobile Connect
4. Aside from the <i>Media Resources --> Mobile Voice Access</i> where else do you need to specify the the Mobile Voice Access DN?
Via the <i>voice hunt-group parallel</i> command
Service Parameters --> <i>Mobile Voice Access Number</i>
It's best to strip digits at the voice port.
5. How do you test a Voice Translation Rule?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
The <i>name</i> configuration field in ephone-dn and voice register dn
#test voice translation rule 1 <input to test>
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.
6. Calls coming from CUCM to PSTN need what?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Outbound dial-peers
It's best to strip digits at the voice port.
7. How do you get around relying on DNS for your CUCMs?
Debug isdn q931
On CUCM it's identical to adding an H323 GW.
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 IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
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
Reroute when here is WAN congestion.
Dtmf-relay h245-alpha
SERVICE PARAMETER> Automated Alternate Routing Enable > True
9. How do you set the Call Park Reversion Timer?
On CUCM it's identical to adding an H323 GW.
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Dial-peers
10. Is the order of the MRGs in an MRGL significant?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Outbound dial-peers
Debug ephone moh
11. What are the commands to configure a SIP phone in CUCME?
Bandwidth
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
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
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
12. How do you setup AutoRegistration in CUCM?
Reroute when here is WAN congestion.
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
#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
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
13. On an H323 GW - how do you adjust the timers for redundancy hunting?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Service Parameters --> <i>Mobile Voice Access Number</i>
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
14. 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?
Call forwarding between voip to voip (when CUBE is in play)
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
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
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.
15. How do you create a trunk on the router side?
Call forwarding between voip to voip (when CUBE is in play)
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
'Use before the ... so XXXX
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
16. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Master/Slave relationship. CUCM controls it.
17. In Gatekeeper CAC how do you restrict a specific endpoint?
...
Show ccm-manager
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Reroute when there is a WAN Outage.
18. What is the full E164 format?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
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
Bottom up - to prevent glare.
19. How do you use an ephone template?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Apply it to the ephone.
20. How do you prioritize route groups?
Allows you to transfer by only pressing the Transfer button once.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Automatically configures the MGCP GW for you.
Put them in a route list.
21. How do you prevent toll fraud on CME?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
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
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 ....
(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
22. How do you see multicast packets being sent?
IP Voice Media Streaming App
Debug ephone moh
'Service Parameters --> <i>Auto Answer Timer</i>
Sends the Calling Name.
23. What types of digit manipulation can you perform at the route pattern?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Device Pool Locations
24. How do you enable AAR system wide?
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.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
24kb/s
Debug ephone moh
25. How do you allow the Calling Name to be sent to the PSTN on a router?
Dtmf-relay h245-alpha
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
26. What does <i>ccm-manager config server [IP]</ip> do?
Sh ccm-manager
Single Number Reach (Mobile Connect)
Automatically configures the MGCP GW for you.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
27. How do you place SCCP and SIP phones into a single huntgroup?
SIP Dial Rules
Via the <i>voice hunt-group parallel</i> command
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
(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
28. Describe how you configure SIP URI functionality.
Higher
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
The <i>name</i> configuration field in ephone-dn and voice register dn
29. How do you support multiple codecs on a dialpeer?
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
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>
Ccm-manager music-on-hold
30. Describe the relationship between route patterns and end devices.
Apply it to the ephone.
<i>Auto Call Pickup Enabled</i>
Outbound 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.
31. <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
...
The Called/Calling Transformations are superceded.
When configuring TEHO.
32. When do you do digit manipulation at the route pattern as opposed to the route list?
#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
When multiple sites use the same route pattern - do your digit manipulation on a route list.
<i>#after-hours block pattern</i>
33. When configuring a Gatekeep-controlled trunk - what do you do with the <i>Wait for Far End H.245 TCS</i> option?
#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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Higher
34. How much bandwidth does a G.729 call including layer 3 require?
Mobile Connect
24kb/s
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
The Called/Calling Transformations are superceded.
35. When using GK redundancy - what must you do to ensure both SUB and PUB use port 1720 for call-signalling.
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36. What's the difference between AAR and SRST?
Outbound dial-peers
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
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
Mobile Connect
37. How do you configure phone ports on a 3750?
Show ccm-manager
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
The <i>name</i> configuration field in ephone-dn and voice register dn
When they are explicitly matched in a destination-pattern in a dial-peer.
38. How do you set up redundancy on outbound dial-peers on an H323 gateway?
'Service Parameters --> <i>Auto Answer Timer</i>
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Debug isdn q931
#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
39. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
When they are explicitly matched in a destination-pattern in a dial-peer.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
40. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
41. Which takes precedence Device Locations or Device Pool Locations?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Assign SIP Dial Rules
The <i>name</i> configuration field in ephone-dn and voice register dn
Device Pool Locations
42. What is call-forward pattern used for?
Call forwarding between voip to voip (when CUBE is in play)
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
#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
Dtmf-relay h245-alpha
43. What is CSIM?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Call Simulator. You can use to validate path from router to the PSTN.
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
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.
44. Privacy is enabled system-wide in CUCM by default. (T or F)
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
TRUE
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
Higher
45. How do you prevent H323 caller-id updates to CUCM
...
#voice service voip| #no supplementary-service h225-notify cid-update
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
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.
46. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
Service Parameters --> <i>Mobile Voice Access Number</i>
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>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
47. How do you configure an MGCP GW? (router side)
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
<i>voice register pool</i>|and|<i>voice register dn</i>
(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
Matches any length dialed number and truncates it to 4 digits.
48. What are the commands to manually configure an MGCP gateway?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
#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
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
49. How do you use the # as a string terminator within a SIP Dial Rule?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Precede the # with a > ... so 9011*>#
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
50. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
#test voice translation rule 1 <input to test>
Sh ip rsvp reservation||sh sccp connections
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
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