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Test your basic knowledge |
CCIE Voice Test
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Study First
Subjects
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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 dtmf-relay type do you use for an H323 GW?
Dtmf-relay h245-alpha
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
On CUCM it's identical to adding an H323 GW.
2. How do you prevent toll fraud on 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
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
The <i>name</i> configuration field in ephone-dn and voice register dn
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
3. Describe how you configure SIP URI functionality.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
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
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
4. How is NTP sych setup in CUCM?
(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
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Configure it on the route group through the route list - then it will be local to the route list.
CUCM OS Administration Settings --> NTP Servers
5. What are the commands to configure a SIP phone in CUCME?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Outbound dial-peers
<i>SIP Route Pattern</i> over a SIP Trunk.
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
6. What is call-forward pattern used for?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
#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
Call forwarding between voip to voip (when CUBE is in play)
7. 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
Put them in a route list.
Sh ccm-manager
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
8. Where do you use VIA zone?
When they are explicitly matched in a destination-pattern in a 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
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
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
9. A <i>show isdn status</i> shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
#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-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Configure it on the route group through the route list - then it will be local to the route list.
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
10. What are the 3 mandatory commands within call-manager-fallback?
Max-dn|max-ephone|ip source-address
TRUE
CUCM OS Administration Settings --> NTP Servers
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
11. How do do configure TEHO?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
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
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
Automatically configures the MGCP GW for you.
12. How do you configure AAR?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Ccm-manager music-on-hold
Master/Slave relationship. CUCM controls it.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
13. In Gatekeeper CAC how do you restrict a specific endpoint?
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
Precede the # with a > ... so 9011*>#
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Dial-peers
14. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
To enable two-stage dialing.
On CUCM it's identical to adding an H323 GW.
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
Apply <i>voice-class codec</i>
15. What are the commands to create vlans on an ESW?
<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 ....
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Dial-peers
16. On an MGCP GW - how could you see the primary and backup CUCM servers?
Show ccm-manager
Apply it to the ephone.
The <i>name</i> configuration field in ephone-dn and voice register dn
17. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
The Called/Calling Transformations are superceded.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
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
18. When configuring a Gatekeep-controlled trunk - what do you do with the <i>Wait for Far End H.245 TCS</i> option?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
<i>Auto Call Pickup Enabled</i>
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
19. How do you set the inter-digit timeout for SIP phones in CME?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Single Number Reach (Mobile Connect)
(config)#voice register dialplan
IP Voice Media Streaming App
20. Which takes precedence Device Locations or Device Pool Locations?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Reroute when here is WAN congestion.
Device Pool Locations
21. How do you set up redundancy on outbound dial-peers on an H323 gateway?
<i>Auto Call Pickup Enabled</i>
Reroute when here is WAN congestion.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Assign SIP Dial Rules
22. How do you ensure that G.711 only is used?
On CUCM it's identical to adding an H323 GW.
...
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
23. How do you use the # as a string terminator within a SIP Dial Rule?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
At the CLI: <i>utils ntp status</i>
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Precede the # with a > ... so 9011*>#
24. Two useful troubleshooting commands for CUCME?
#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
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
CUCM OS Administration Settings --> NTP Servers
25. When do you do digit manipulation at the route pattern as opposed to the route list?
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
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.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
26. What does the <i>Transfer On-hook Enabled</i> Service Parameter do?
#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
Allows you to transfer by only pressing the Transfer button once.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
27. How do you enable AAR?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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
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
When they are explicitly matched in a destination-pattern in a dial-peer.
28. What are two caveats to using the <i>ccm-manager config server</ip> command?
#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
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
...
29. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
...
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
(config)#voice register dialplan
To enable two-stage dialing.
30. Where can you assign the AAR Group?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Debug ephone moh
#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
31. What are the commands to configure an H323 GW?
#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
24kb/s
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
#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
32. How do you support multiple codecs on a dialpeer?
#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 ... this is not optional!
Apply <i>voice-class codec</i>
Matches any length dialed number and truncates it to 4 digits.
33. 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.
To enable two-stage dialing.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
34. Privacy is enabled system-wide in CUCM by default. (T or F)
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
TRUE
(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
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
35. In CME Where is the Calling Name derived from?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Single Number Reach (Mobile Connect)
The <i>name</i> configuration field in ephone-dn and voice register dn
Call forwarding between voip to voip (when CUBE is in play)
36. When are digits stripped in a 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
SERVICE PARAMETER> Automated Alternate Routing Enable > True
When they are explicitly matched in a destination-pattern in a dial-peer.
The <i>name</i> configuration field in ephone-dn and voice register dn
37. How do you allow H323 calls to be preserved should the primary H323 GW fail?
To notify SIP Phones of NTP
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.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
38. How do you enable Extension Mobility for a device in CME?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
To notify SIP Phones of NTP
When they are explicitly matched in a destination-pattern in a dial-peer.
Assign a logout-profile to the ephone.
39. What are the commands to configure SRST in fallback?
Device Pool Locations
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
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
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
40. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Ccm-manager music-on-hold
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
TRUE
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
41. How do you see multicast packets being sent?
Debug ephone moh
1 ... this is not optional!
Single Number Reach (Mobile Connect)
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
42. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
When configuring TEHO.
43. What are the commands to set up the PRI?
44. How do you prioritize route groups?
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.
Show ccm-manager
Allows you to transfer by only pressing the Transfer button once.
Put them in a route list.
45. How do you create a trunk on the switch 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
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
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. What is the bit rate for a G.729 call excluding layer 2?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
8kb/s
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
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
47. What terminology translates to AAR?
When they are explicitly matched in a destination-pattern in a dial-peer.
Reroute when here is WAN congestion.
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>
48. How do you see the details of calls coming in and out of the PRI?
'Use before the ... so XXXX
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Debug isdn q931
49. How do you change modes in <i>voice register global</i>?
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)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
#test voice translation rule 1 <input to test>
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
50. How do you prevent H323 caller-id updates to CUCM
Top Down means that channel 1 will be the first channel used to place outgoing calls.
SIP Dial Rules
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
#voice service voip| #no supplementary-service h225-notify cid-update