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Test your basic knowledge |
CCIE Voice Test
Start Test
Study First
Subjects
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cisco
,
it-skills
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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. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
<i>SIP Route Pattern</i> 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
<i>voice register pool</i>|and|<i>voice register dn</i>
2. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
To enable two-stage dialing.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
3. A <i>show isdn status</i> shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Device Pool Locations
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
4. 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
(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
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Bottom up - to prevent glare.
5. What are the commands to manually configure an MGCP gateway?
#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
Assign SIP Dial Rules
The Called/Calling Transformations are superceded.
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
6. How do you configure phone ports on an ESW?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Call Simulator. You can use to validate path from router to the PSTN.
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
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
7. When are digits stripped in a gateway?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
SIP Dial Rules
Mobile Voice Access
When they are explicitly matched in a destination-pattern in a dial-peer.
8. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
Mobile Voice Access
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
When multiple sites use the same route pattern - do your digit manipulation on a route list.
9. What's the best way to do digit manipulation on an IOS gateway?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
24kb/s
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
10. What are the commands to configure SRST in fallback?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
<i>SIP Route Pattern</i> over a SIP Trunk.
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
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
11. Is the + character supported on a VOIP dial-peer?
12. How do you configure CUBE?
13. What are the commands to configure a T1/E1 PRI?
Push as much digit manipulation to the router as possible - so you can re-use it for 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
(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
Sh ccm-manager
14. How do you ensure that G.711 only is used?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
#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
<i>Auto Call Pickup Enabled</i>
15. What are the base telephony-service commands for CME?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
16. How can you confirm the MGCP GW is registered to CUCM - in IOS?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Sh ccm-manager
17. How do you configure SRST?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Precede the # with a > ... so 9011*>#
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
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
18. How do you setup AutoRegistration in CUCM?
<i>voice register pool</i>|and|<i>voice register dn</i>
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.
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
(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
19. How do you set the inter-digit timeout for SIP phones in CME?
(config)#voice register dialplan
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Put them in a route list.
Bandwidth
20. What is call-forward pattern used for?
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
Call forwarding between voip to voip (when CUBE is in play)
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
Higher
21. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Reroute when there is a WAN Outage.
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
22. How is NTP sych setup in CUCM?
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
CUCM OS Administration Settings --> NTP Servers
23. How do you see the details of calls coming in and out of the PRI?
Debug isdn q931
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
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 .)
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
24. How do you use the # as a string terminator within a SIP Dial Rule?
Precede the # with a > ... so 9011*>#
Call forwarding between voip to voip (when CUBE is in play)
Assign SIP Dial Rules
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
25. Name 4 useful show commands for active calls.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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
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 .)
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
26. What are the commands to configure NTP in IOS?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
#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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
27. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Automatically configures the MGCP GW for you.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Apply it to the ephone.
28. What does KPML do?
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.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
29. How do you prioritize route groups?
Put them in a route list.
The Called/Calling Transformations are superceded.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Ccm-manager music-on-hold
30. What is the bit rate for a G.729 call excluding layer 2?
8kb/s
Matches any length dialed number and truncates it to 4 digits.
At the CLI: <i>utils ntp status</i>
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
31. How do you prevent toll fraud on CME?
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 ....
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
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
'Service Parameters --> <i>Auto Answer Timer</i>
32. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
IP Voice Media Streaming App
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
33. How do you verify where MOH is being served up from?
34. What commands are needed to configure the voice register pool in CME?
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
CUCM OS Administration Settings --> NTP Servers
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
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
35. An MGCP gateway serving as an SRST router requires what?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
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.
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
36. How do you enable Extension Mobility for a device in CME?
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
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Assign a logout-profile to the ephone.
8kb/s
37. Maximum Wait Time for Desk Pickup?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
#voice service voip| #no supplementary-service h225-notify cid-update
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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
38. How do you place SCCP and SIP phones into a single huntgroup?
Via the <i>voice hunt-group parallel</i> command
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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
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.
39. What would force you to use telephony-service to configure SRST?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
1 ... this is not optional!
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. What are the steps to configure Single Number Reach and Mobile Voice Access?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
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
41. What are the basic SCCP Commands fro telephony-service in CUCME?
(config)#voice register dialplan
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
42. What are the commands to configure an H323 GW?
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.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
#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
#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
43. How do you use an ephone template?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
#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
Apply it to the ephone.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
44. How do you allow the Calling Name to be sent to the PSTN on a router?
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
To notify SIP Phones of NTP
On CUCM it's identical to adding an H323 GW.
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
45. Where is it best to manipulate digits for inbound calls?
46. Describe how you configure SIP URI functionality.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
TRUE
Debug ephone moh
47. How do you create a trunk on the router side?
To enable two-stage dialing.
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
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
48. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
Matches any length dialed number and truncates it to 4 digits.
Device Pool Locations
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
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
49. What does <i>ccm-manager config server [IP]</ip> do?
Assign a logout-profile to the ephone.
(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
Automatically configures the MGCP GW for you.
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
50. '<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
...
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
The Called/Calling Transformations are superceded.
Mobile Voice Access