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Test your basic knowledge |
CCIE Voice Test
Start Test
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. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?
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2. How do you create a trunk on the switch side?
Mobile Connect
CUCM OS Administration Settings --> NTP Servers
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
(config)#<i>sh cdp neigh detail
3. What's the difference between AAR and SRST?
SIP Dial Rules
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
#voice service voip| #no supplementary-service h225-notify cid-update
'Service Parameters --> <i>Auto Answer Timer</i>
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
#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
Mobile Connect
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
5. How do do configure TEHO?
Apply it to the ephone.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
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
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
6. What is CSIM?
Call Simulator. You can use to validate path from router to the PSTN.
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
Sh ccm-manager
...
7. What would force you to use telephony-service to configure SRST?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Precede the # with a > ... so 9011*>#
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)
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
8. How do you create a trunk on the router side?
<i>#after-hours block pattern</i>
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)
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.
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
9. How do you configure DHCP in IOS?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
(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
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
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
10. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
Device Pool Locations
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
...
11. When setting up SIP URI where do you configure the CUCM's domain name?'
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Debug ephone moh
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
Via the <i>voice hunt-group parallel</i> command
12. How do you enable AAR system wide?
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
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 ....
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Reroute when there is a WAN Outage.
13. How do you enable AAR?
Single Number Reach (Mobile Connect)
(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
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Matches any length dialed number and truncates it to 4 digits.
14. What are the commands to configure SRST in fallback?
Mobile Voice Access
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
(config)#<i>sh cdp neigh detail
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
15. How do you configure CUBE?
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16. What kind of relationship does an MGCP gateway have with CUCM?
Master/Slave relationship. CUCM controls it.
#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
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 <i>name</i> configuration field in ephone-dn and voice register dn
17. How do you configure an MGCP GW? (router side)
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
Debug ephone moh
(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
18. What are the commands to set up the PRI?
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19. Privacy is enabled system-wide in CUCM by default. (T or F)
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
TRUE
Dtmf-relay h245-alpha
Higher
20. How do you test a Voice Translation Rule?
Ccm-manager music-on-hold
Assign a logout-profile to the ephone.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
#test voice translation rule 1 <input to test>
21. What are the commands to configure a MGCP Gateway? (router)
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
(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
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
22. What is the bit rate for a G.729 call excluding layer 2?
8kb/s
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Matches any length dialed number and truncates it to 4 digits.
23. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
<i>SIP Route Pattern</i> over a SIP Trunk.
TRUE
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.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
24. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
Reroute when here is WAN congestion.
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
To enable two-stage dialing.
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
25. How do you ensure that G.711 only is used?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
Assign a logout-profile to the ephone.
CUCM OS Administration Settings --> NTP Servers
26. When a CUCM device dials a number - what happens?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
CUCM OS Administration Settings --> NTP Servers
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
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 .)
27. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
Reroute when here is WAN congestion.
<i>voice register pool</i>|and|<i>voice register dn</i>
Automatically configures the MGCP GW for you.
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
28. What terminology translates to AAR?
Reroute when here is WAN congestion.
On CUCM it's identical to adding an H323 GW.
(config)#voice register dialplan
29. If you're using Called/Calling Party Transformations at the Route Pattern and tranformations at the route list ... What does the RP affect and What does the RL affect?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
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 ....
30. What are the steps to configure Single Number Reach and Mobile Voice Access?
(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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
31. 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?
(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
#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
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
32. 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?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Precede the # with a > ... so 9011*>#
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.
#voice service voip| #no supplementary-service h225-notify cid-update
33. What does KPML do?
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
(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
34. What should you configure before entering auto qos?
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.
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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
35. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Put them in a route list.
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 Parameters --> <i>Auto Answer Timer</i>
Bottom up - to prevent glare.
36. How do you use the # as a string terminator within a SIP Dial Rule?
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.
Precede the # with a > ... so 9011*>#
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Apply <i>voice-class codec</i>
37. How do you configure a SIP Trunk? (router side)
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
(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
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
38. What are the commands to configure NTP in IOS?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
39. What dtmf-relay type do you use for an H323 GW?
Reroute when there is a WAN Outage.
Dtmf-relay h245-alpha
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
40. SNR is also known as?
#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
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
Mobile Connect
(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
41. What does Display-IE 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.
Sends the Calling Name.
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
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
42. Maximum Wait Time for Desk Pickup?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
TRUE
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
43. 'How do you inform a SIP phone of NTP information?
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
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 Voice Access
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
44. How do you enable Extension Mobility for a device in CME?
Reroute when here is WAN congestion.
Assign a logout-profile to the ephone.
Dtmf-relay h245-alpha
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)
45. Which takes precedence Device Locations or Device Pool Locations?
To enable two-stage dialing.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Via the <i>voice hunt-group parallel</i> command
Device Pool Locations
46. Name 4 useful show commands for active calls.
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.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Max-dn|max-ephone|ip source-address
47. What does the <i>Transfer On-hook Enabled</i> Service Parameter do?
Allows you to transfer by only pressing the Transfer button once.
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
Put them in a route list.
Ccm-manager music-on-hold
48. What are the basic ephone-dn and ephone commands?
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
...
49. 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
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Matches any length dialed number and truncates it to 4 digits.
50. Name 2 commands to verify RSVP functionality.
1 ... this is not optional!
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.
Sh ip rsvp reservation||sh sccp connections
#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