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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. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
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
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
2. What types of digit manipulation can you perform at the route pattern?
(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
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
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 .)
3. 'How do you inform a SIP phone of NTP information?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
4. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
24kb/s
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
5. Two useful troubleshooting commands for CUCME?
Debug ephone moh
#test voice translation rule 1 <input to test>
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
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
6. How do you configure a gateway to register with gatekeeper?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
1 ... this is not optional!
Master/Slave relationship. CUCM controls it.
#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
7. How do you enable Extension Mobility for a device in CME?
Assign a logout-profile to the ephone.
<i>#after-hours block pattern</i>
Reroute when there is a WAN Outage.
'Use before the ... so XXXX
8. How would you enable security on a GK?
When they are explicitly matched in a destination-pattern in a dial-peer.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
#voice service voip| #no supplementary-service h225-notify cid-update
9. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
Service Parameters --> <i>Mobile Voice Access Number</i>
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
SIP Dial Rules
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
10. On the RDP What is the CSS used for?
#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
Mobile Voice Access
It's best to strip digits at the voice port.
Call Simulator. You can use to validate path from router to the PSTN.
11. 'If you want to make changes to any softkeys where do you do it?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
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
12. <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 os truly unsurpassed. I have opened up a wealth of opportunities for myself and my family and
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
...
When configuring TEHO.
13. How do you set up redundancy on outbound dial-peers on an H323 gateway?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
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
1 ... this is not optional!
14. What is the bit rate for a G.729 call excluding layer 2?
Sends the Calling Name.
Outbound dial-peers
8kb/s
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
15. How do you set the Call Park Reversion Timer?
Single Number Reach (Mobile Connect)
'Use before the ... so XXXX
#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
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
16. What CUCM services should you activate?
Sh ccm-manager
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
17. 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
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
18. On the RDP What is the Rerouting CSS used for?
Device Pool Locations
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Single Number Reach (Mobile Connect)
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
19. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
Allows you to transfer by only pressing the Transfer button once.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Top Down means that channel 1 will be the first channel used to place outgoing calls.
20. Privacy is enabled system-wide in CUCM by default. (T or F)
TRUE
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
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
21. A <i>show isdn status</i> shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Sh ip rsvp reservation||sh sccp connections
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Allows you to transfer by only pressing the Transfer button once.
22. How do you prevent toll fraud on CUCM?
#test voice translation rule 1 <input to test>
SIP Dial Rules
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
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
23. Calls coming from CUCM to PSTN need what?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
To notify SIP Phones of NTP
Outbound dial-peers
24. How do you set the inter-digit timeout for SIP phones in CME?
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.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
(config)#voice register dialplan
Put them in a route list.
25. What would force you to use telephony-service to configure SRST?
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Bottom up - to prevent glare.
To notify SIP Phones of NTP
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
26. Aside from the <i>Media Resources --> Mobile Voice Access</i> where else do you need to specify the the Mobile Voice Access DN?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Service Parameters --> <i>Mobile Voice Access Number</i>
<i>voice register pool</i>|and|<i>voice register dn</i>
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
27. What must you do for a BACD script to work on a CME router?
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
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Automatically configures the MGCP GW for you.
28. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
<i>SIP Route Pattern</i> over a SIP Trunk.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Assign a logout-profile to the ephone.
29. How do you enable AAR?
It's best to strip digits at the voice port.
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
30. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
Put them in a route list.
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Reroute when here is WAN congestion.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
31. What dtmf-relay type do you use for an H323 GW?
Via the <i>voice hunt-group parallel</i> command
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 PARAMETER> Automated Alternate Routing Enable > True
Dtmf-relay h245-alpha
32. How do you configure SRST?
Call-manager-fallback</i> or <i>telephony-service mode SRST</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.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
33. 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
#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
Dtmf-relay h245-alpha
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
34. What are the basic SCCP Commands fro telephony-service in CUCME?
<i>#after-hours block pattern</i>
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
35. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
Sh ccm-manager
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.
<i>Auto Call Pickup Enabled</i>
(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
36. What's the best way to perform digit manipulation on a route group?
(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
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.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
37. How do you use an ephone template?
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
Apply it to the ephone.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
38. How is NTP sych setup in CUCM?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
CUCM OS Administration Settings --> NTP Servers
Ccm-manager music-on-hold
39. With the gw-priority command - does higher or lower priority take precedence?
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
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Higher
40. When are digits stripped in a gateway?
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
When they are explicitly matched in a destination-pattern in a dial-peer.
Dtmf-relay h245-alpha
#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
41. What are the 3 mandatory commands within call-manager-fallback?
Max-dn|max-ephone|ip source-address
#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
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
On CUCM it's identical to adding an H323 GW.
42. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
When they are explicitly matched in a destination-pattern in a dial-peer.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
...
43. What are the commands to configure a SIP phone in CUCME?
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
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
Configure it on the route group through the route list - then it will be local to the route list.
#test voice translation rule 1 <input to test>
44. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
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
Matches any length dialed number and truncates it to 4 digits.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
Bottom up - to prevent glare.
45. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
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.
#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>
46. What is SIP URI?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Sh ip rsvp reservation||sh sccp connections
47. How do you use the # as a string terminator within a SIP Dial Rule?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Precede the # with a > ... so 9011*>#
The Called/Calling Transformations are superceded.
48. How do you change modes in <i>voice register global</i>?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
49. How do you prioritize route groups?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Put them in a route list.
<i>SIP Route Pattern</i> over a SIP Trunk.
Assign SIP Dial Rules
50. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
1 ... this is not optional!
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 ....
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>