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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. What are the commands to configure SRST in fallback?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
2. When can you not use a Standard Local Route Group?
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
When configuring TEHO.
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
3. How do you configure a gateway to register with gatekeeper?
#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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Sh ccm-manager
4. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
5. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
<i>voice register pool</i>|and|<i>voice register dn</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
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Apply <i>voice-class codec</i>
6. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
Bandwidth
Call forwarding between voip to voip (when CUBE is in play)
Call Simulator. You can use to validate path from router to the PSTN.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
7. How do you disable KPML?
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 SIP Dial Rules
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
8. 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>
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
SIP Dial Rules
9. How is NTP sych setup in CUCM?
Precede the # with a > ... so 9011*>#
CUCM OS Administration Settings --> NTP Servers
#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
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)
10. How do you use the # as a string terminator within a SIP Dial Rule?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Precede the # with a > ... so 9011*>#
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
11. What are the commands to configure NTP in IOS?
Precede the # with a > ... so 9011*>#
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Assign a logout-profile to the ephone.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
12. What are the commands to configure a T1/E1 PRI?
Debug isdn q931
(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
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
13. What is call-forward pattern used for?
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
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
When configuring TEHO.
Call forwarding between voip to voip (when CUBE is in play)
14. How do you configure class of service (CoS) in CUCM? CME?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
#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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
15. What must you do for a BACD script to work on a CME router?
Allows you to transfer by only pressing the Transfer button once.
Bandwidth
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
16. What CM Service needs to be start in Serviceability for MOH to work?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
IP Voice Media Streaming App
Apply it to the ephone.
17. Where do you use VIA zone?
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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
(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
Higher
18. What is SIP URI?
In CUCM - configure CFUR to point to it's E164 number.
When configuring TEHO.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
19. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
...
#test voice translation rule 1 <input to test>
<i>SIP Route Pattern</i> over a SIP Trunk.
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
20. What kind of relationship does an MGCP gateway have with CUCM?
Master/Slave relationship. CUCM controls it.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_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 .)
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
21. 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 .)
Bottom up - to prevent glare.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
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
22. <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
'Service Parameters --> <i>Auto Answer Timer</i>
...
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
23. What do you need to do to activate the CME GUI?
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
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
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
(config)#voice register dialplan
24. On the RDP What is the CSS used for?
Mobile Voice Access
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Device Pool Locations
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
25. What should you configure before entering auto qos?
To notify SIP Phones of NTP
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
Bandwidth
26. How do you configure DHCP in IOS?
The <i>name</i> configuration field in ephone-dn and voice register dn
(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
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
#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
27. When setting up SIP URI where do you configure the CUCM's domain name?'
To enable two-stage dialing.
Matches any length dialed number and truncates it to 4 digits.
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
28. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
The Called/Calling Transformations are superceded.
TRUE
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
29. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Higher
Device Pool Locations
Need to Know||Anytime you have an external call - it needs to match on a 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
30. How do you configure a SIP Trunk? (router side)
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)
(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 the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
31. Where is it best to manipulate digits for inbound calls?
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32. When using GK redundancy - what must you do to ensure both SUB and PUB use port 1720 for call-signalling.
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33. In Gatekeeper CAC how do you restrict a specific endpoint?
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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Master/Slave relationship. CUCM controls it.
34. What are the base telephony-service commands for CME?
It's best to strip digits at the voice port.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
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
#voice service voip| #no supplementary-service h225-notify cid-update
35. What terminology translates to SRST?
Reroute when there is a WAN Outage.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
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
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
36. What does <i>ccm-manager config server [IP]</ip> do?
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 ....
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
To enable two-stage dialing.
Automatically configures the MGCP GW for you.
37. Is the order of the MRGs in an MRGL significant?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
(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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
Apply <i>voice-class codec</i>
38. What would force you to use telephony-service to configure SRST?
Assign a logout-profile to the ephone.
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.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
39. Does CUCM support RSVP natively?
#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
#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
When they are explicitly matched in a destination-pattern in a dial-peer.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
40. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
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.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
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.
41. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
Call Simulator. You can use to validate path from router to the PSTN.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Via the <i>voice hunt-group parallel</i> command
Max-dn|max-ephone|ip source-address
42. What is the bit rate for a G.729 call excluding layer 2?
8kb/s
Precede the # with a > ... so 9011*>#
Call-manager-fallback</i> or <i>telephony-service mode SRST</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
43. What does KPML do?
8kb/s
SERVICE PARAMETER> Automated Alternate Routing Enable > True
Debug ephone moh
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
44. 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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Allows you to transfer by only pressing the Transfer button once.
45. How do you allow the Calling Name to be sent to the PSTN on a router?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
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.
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
CUCM OS Administration Settings --> NTP Servers
46. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
Single Number Reach (Mobile Connect)
1 ... this is not optional!
Matches any length dialed number and truncates it to 4 digits.
Dtmf-relay h245-alpha
47. How do you set up redundancy on outbound dial-peers on an H323 gateway?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Apply it to the ephone.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
48. How do you create a trunk on the switch side?
Assign SIP Dial Rules
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
49. How do you ensure that G.711 only is used?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
The Called/Calling Transformations are superceded.
Call forwarding between voip to voip (when CUBE is in play)
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
50. What terminology translates to AAR?
Reroute when here is WAN congestion.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
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
Mobile Voice Access