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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. 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
Assign SIP Dial Rules
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
(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
2. In Gatekeeper CAC how do you restrict a specific endpoint?
Reroute when there is a WAN Outage.
It's best to strip digits at the voice port.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
3. How do you verify that NTP is working on the CUCM server?
'Use before the ... so XXXX
Sends the Calling Name.
At the CLI: <i>utils ntp status</i>
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
4. <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
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
...
5. When can you not use a Standard Local Route Group?
When configuring TEHO.
To notify SIP Phones of NTP
Dial-peers
6. How do you use the # as a string terminator within a SIP Dial Rule?
Precede the # with a > ... so 9011*>#
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
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.
7. What kind of relationship does an MGCP gateway have with CUCM?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
<i>voice register pool</i>|and|<i>voice register dn</i>
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
Master/Slave relationship. CUCM controls it.
8. 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>
SIP Dial Rules
#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>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
9. What terminology translates to AAR?
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
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Reroute when here is WAN congestion.
Debug isdn q931
10. Which takes precedence Device Locations or Device Pool Locations?
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Device Pool Locations
11. When setting up SIP URI where do you configure the CUCM's domain name?'
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
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
24kb/s
12. How do you set up redundancy on outbound dial-peers on an H323 gateway?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
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)
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
(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
13. Privacy is enabled system-wide in CUCM by default. (T or F)
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Mobile Connect
TRUE
14. What CM Service needs to be start in Serviceability for MOH to work?
IP Voice Media Streaming App
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
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
15. With the gw-priority command - does higher or lower priority take precedence?
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)
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Higher
<i>#after-hours block pattern</i>
16. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
SIP Dial Rules
Allows you to transfer by only pressing the Transfer button once.
On CUCM it's identical to adding an H323 GW.
Bottom up - to prevent glare.
17. How do you test a Voice Translation Rule?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
#test voice translation rule 1 <input to test>
Apply it to the ephone.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
18. What does <i>ccm-manager config server [IP]</ip> do?
Automatically configures the MGCP GW for you.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
Top Down means that channel 1 will be the first channel used to place outgoing calls.
#test voice translation rule 1 <input to test>
19. How do you configure AAR?
Call forwarding between voip to voip (when CUBE is in play)
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
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
20. What are the steps to integrate CUPS with 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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Via the <i>voice hunt-group parallel</i> command
21. What are the commands to configure a MGCP Gateway? (router)
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
22. What CUCM services should you activate?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
...
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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 .)
23. What are the basic SCCP Commands fro telephony-service in CUCME?
#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
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
When they are explicitly matched in a destination-pattern in a dial-peer.
24. On the RDP What is the Rerouting CSS used for?
Automatically configures the MGCP GW for you.
<i>Auto Call Pickup Enabled</i>
Single Number Reach (Mobile Connect)
#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
25. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
Matches any length dialed number and truncates it to 4 digits.
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.
8kb/s
<i>#after-hours block pattern</i>
26. When a CUCM device dials a number - what happens?
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 .)
To notify SIP Phones of NTP
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
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
27. How do you configure phone ports on a 3750?
Debug isdn q931
IP Voice Media Streaming App
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
28. How do you configure CUBE?
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29. What must you do for a BACD script to work on a CME router?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
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
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
Matches any length dialed number and truncates it to 4 digits.
30. How would you set the the timer for Auto Answer?
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31. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
SIP Dial Rules
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
32. How do you enable AAR?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Reroute when there is a WAN Outage.
33. How do you see the details of calls coming in and out of the PRI?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Debug isdn q931
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
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
34. How do you allow the Calling Name to be sent to the PSTN on a router?
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)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
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>voice register pool</i>|and|<i>voice register dn</i>
35. Where is it best to manipulate digits for inbound calls?
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36. Name 4 useful show commands for active calls.
The <i>name</i> configuration field in ephone-dn and voice register dn
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Via the <i>voice hunt-group parallel</i> command
37. How would you verify that DHCP is working in IOS?
(config)#<i>sh cdp neigh detail
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
Higher
Precede the # with a > ... so 9011*>#
38. What do you need to do to activate the CME GUI?
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
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
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
39. How do you configure a SIP Trunk? (router side)
(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
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Apply <i>voice-class codec</i>
TRUE
40. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
<i>Auto Call Pickup Enabled</i>
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Dial-peers
41. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
42. How do you configure DHCP in IOS?
To enable two-stage dialing.
(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
The Called/Calling Transformations are superceded.
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.
43. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
Max-dn|max-ephone|ip source-address
#voice service voip| #no supplementary-service h225-notify cid-update
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
44. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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.
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
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.
Higher
45. Calls coming from CUCM to PSTN need what?
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
Outbound dial-peers
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
46. What types of digit manipulation can you perform at the route pattern?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
47. How do you prevent H323 caller-id updates to CUCM
#voice service voip| #no supplementary-service h225-notify cid-update
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
Single Number Reach (Mobile Connect)
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
48. If you can't use a tech-prefix or default tech-prefix how do you get around the fact that CUCM does not register it's endpoint to GK?
#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
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
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)
49. How do you block an external call from being transferred back out to the pstn by an internal user?
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
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
Single Number Reach (Mobile Connect)
SERVICE PARAMETER> Automated Alternate Routing Enable > True
50. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
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
...
The Called/Calling Transformations are superceded.