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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 commands are needed to configure the voice register pool in CME?
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
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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
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.
2. How do you configure SRST?
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
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
3. Is the + character supported on a VOIP dial-peer?
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4. When using <i>drop-through-option</i> What is the max number of huntgroups?
Assign a logout-profile to the ephone.
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
Service Parameters --> <i>Mobile Voice Access Number</i>
1 ... this is not optional!
5. 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?
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)
Call Simulator. You can use to validate path from router to the PSTN.
Assign SIP Dial Rules
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
6. <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
Bandwidth
Mobile Connect
...
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
7. How do you see multicast packets being sent?
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
#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
Debug ephone moh
Debug isdn q931
8. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
'Service Parameters --> <i>Auto Answer Timer</i>
Max-dn|max-ephone|ip source-address
Matches any length dialed number and truncates it to 4 digits.
9. What are the commands to configure NTP in IOS?
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
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
10. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Dtmf-relay h245-alpha
...
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
11. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
Ccm-manager music-on-hold
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</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
12. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
<i>Auto Call Pickup Enabled</i>
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
13. What terminology translates to SRST?
Reroute when there is a WAN Outage.
Allows you to transfer by only pressing the Transfer button once.
Call forwarding between voip to voip (when CUBE is in play)
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
14. When are digits stripped in a gateway?
When they are explicitly matched in a destination-pattern in a dial-peer.
#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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
15. 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.
<i>voice translation-profile</i>|and|<i>voice translation-rule</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
When configuring TEHO.
16. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?
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17. What are the steps to integrate CUPS with CUCM?
...
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Higher
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
18. Describe how you configure SIP URI functionality.
Assign SIP Dial Rules
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
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
19. <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
SIP Dial Rules
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
...
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.
20. What are the commands to configure a MGCP Gateway? (router)
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Call forwarding between voip to voip (when CUBE is in play)
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
#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
21. What are the 3 mandatory commands within call-manager-fallback?
Debug isdn q931
Max-dn|max-ephone|ip source-address
Mobile Voice Access
Higher
22. What are the commands to manually configure an MGCP 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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Assign SIP Dial Rules
The <i>name</i> configuration field in ephone-dn and voice register dn
23. How would you enable security on a GK?
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Reroute when there is a WAN Outage.
Outbound dial-peers
24. What are the commands to create vlans on an ESW?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
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
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
25. How do you configure an MGCP GW? (router side)
Dial-peers
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
(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
26. How do you configure CUCM redundancy on an H323 gateway?
'Use before the ... so XXXX
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
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:(SUB IP)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #voice-clas
27. Calls coming from CUCM to PSTN need what?
The Called/Calling Transformations are superceded.
Outbound dial-peers
IP Voice Media Streaming App
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
28. How do you enable Extension Mobility for a device in CME?
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
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
On CUCM it's identical to adding an H323 GW.
Assign a logout-profile to the ephone.
29. How do you prioritize route groups?
(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
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Put them in a route list.
30. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
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>
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
31. How do you allow the Calling Name to be sent to the PSTN on a router?
Device Pool Locations
(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-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
32. How do you use the # as a string terminator within a SIP Dial Rule?
Precede the # with a > ... so 9011*>#
Ccm-manager music-on-hold
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
Configure it on the route group through the route list - then it will be local to the route list.
33. What is the bit rate for a G.729 call excluding layer 2?
Reroute when there is a WAN Outage.
Show ccm-manager
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
8kb/s
34. What are the base telephony-service commands for CME?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
24kb/s
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
35. How do you prevent toll fraud on CUCM?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
24kb/s
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
36. What CM Service needs to be start in Serviceability for MOH to work?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
IP Voice Media Streaming App
TRUE
...
37. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
<i>voice register pool</i>|and|<i>voice register dn</i>
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
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.
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
38. How is NTP sych setup in CUCM?
CUCM OS Administration Settings --> NTP Servers
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
39. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
Matches any length dialed number and truncates it to 4 digits.
<i>voice register pool</i>|and|<i>voice register dn</i>
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.
Call forwarding between voip to voip (when CUBE is in play)
40. How do you see the details of calls coming in and out of the PRI?
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
When multiple sites use the same route pattern - do your digit manipulation on a route list.
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.
Debug isdn q931
41. What are the commands to configure SIP phones in CME?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
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
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
42. An MGCP gateway serving as an SRST router requires what?
Sends the Calling Name.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Outbound dial-peers
43. How do you configure AAR?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
44. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
(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
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Bottom up - to prevent glare.
45. 'How do you inform a SIP phone of NTP information?
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
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)
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
46. How do you set up redundancy on outbound dial-peers on an H323 gateway?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Apply it to the ephone.
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-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
47. How do you support multiple codecs on a dialpeer?
#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
Apply <i>voice-class codec</i>
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Sh ccm-manager
48. How do you ensure that G.711 only is used?
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
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
Apply it to the ephone.
49. How would you verify that DHCP is working in IOS?
(config)#<i>sh cdp neigh detail
(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
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
50. 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
The Called/Calling Transformations are superceded.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
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