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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. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
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.
2. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
The Called/Calling Transformations are superceded.
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
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
3. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</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
4. What does the <i>Transfer On-hook Enabled</i> Service Parameter do?
To enable two-stage dialing.
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
Allows you to transfer by only pressing the Transfer button once.
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
5. How do you configure class of service (CoS) in CUCM? CME?
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Put them in a route list.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
6. How do you get around relying on DNS for your CUCMs?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
(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
#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
7. How do you prevent H323 caller-id updates to CUCM
#voice service voip| #no supplementary-service h225-notify cid-update
(config)#<i>sh cdp neigh detail
Higher
(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
8. What terminology translates to AAR?
Dial-peers
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
To enable two-stage dialing.
Reroute when here is WAN congestion.
9. 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?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
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)
Apply it to the ephone.
'Use before the ... so XXXX
10. On the RDP What is the CSS used for?
Mobile Voice Access
On CUCM it's identical to adding an H323 GW.
#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
(config)#voice register dialplan
11. How do you configure SRST?
Bottom up - to prevent glare.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
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
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
12. How do you configure DHCP in IOS?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
(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
To notify SIP Phones of NTP
13. How do you allow H323 calls to be preserved should the primary H323 GW fail?
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Allows you to transfer by only pressing the Transfer button once.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
14. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
<i>Auto Call Pickup Enabled</i>
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Debug ephone moh
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
15. How do you use the # as a string terminator within a SIP Dial Rule?
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
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.
Precede the # with a > ... so 9011*>#
16. Describe how you configure SIP URI functionality.
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 .)
Put them in a route list.
Master/Slave relationship. CUCM controls it.
17. What does an H323 GW require that MGCP GWs do not?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Dial-peers
(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
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
18. Aside from the <i>Media Resources --> Mobile Voice Access</i> where else do you need to specify the the Mobile Voice Access DN?
Sh ip rsvp reservation||sh sccp connections
Service Parameters --> <i>Mobile Voice Access Number</i>
Outbound dial-peers
19. Describe the relationship between route patterns and end devices.
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 .)
(config)#<i>sh cdp neigh detail
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.
TRUE
20. An MGCP gateway serving as an SRST router requires what?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
On CUCM it's identical to adding an H323 GW.
<i>#after-hours block pattern</i>
21. How do you change modes in <i>voice register global</i>?
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
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
Apply <i>voice-class codec</i>
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
22. How do you place SCCP and SIP phones into a single huntgroup?
Via the <i>voice hunt-group parallel</i> command
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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
SIP Dial Rules
23. What CM Service needs to be start in Serviceability for MOH to work?
<i>Auto Call Pickup Enabled</i>
Call Simulator. You can use to validate path from router to the PSTN.
IP Voice Media Streaming App
Top Down means that channel 1 will be the first channel used to place outgoing calls.
24. How do you see the details of calls coming in and out of the PRI?
'Use before the ... so XXXX
SIP Dial Rules
Debug isdn q931
When multiple sites use the same route pattern - do your digit manipulation on a route list.
25. Where is it best to manipulate digits for inbound calls?
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on line
183
26. What's the difference between AAR and SRST?
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 .)
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
27. What types of digit manipulation can you perform at the route pattern?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
<i>#after-hours block pattern</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
28. How do you prevent toll fraud on CME?
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 ....
Service Parameters --> <i>Mobile Voice Access Number</i>
<i>voice register pool</i>|and|<i>voice register dn</i>
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
29. <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
It's best to strip digits at the voice port.
...
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
30. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
31. What is CSIM?
Call Simulator. You can use to validate path from router to the PSTN.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Sh ccm-manager
32. What does KPML do?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
33. What are two caveats to using the <i>ccm-manager config server</ip> command?
Precede the # with a > ... so 9011*>#
Mobile Voice Access
When multiple sites use the same route pattern - do your digit manipulation on a route list.
34. How do you use an ephone template?
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
Apply it to the ephone.
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
Outbound dial-peers
35. When using <i>drop-through-option</i> What is the max number of huntgroups?
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.
1 ... this is not optional!
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
36. How would you verify that DHCP is working in IOS?
When they are explicitly matched in a destination-pattern in a dial-peer.
(config)#<i>sh cdp neigh detail
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
37. What are the commands to configure SRST in fallback?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
It's best to strip digits at the voice port.
38. In Gatekeeper CAC how do you restrict a specific endpoint?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
SERVICE PARAMETER> Automated Alternate Routing Enable > True
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
39. What are the commands to configure SIP phones in CME?
<i>Auto Call Pickup Enabled</i>
Single Number Reach (Mobile Connect)
Bottom up - to prevent glare.
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
40. Privacy is enabled system-wide in CUCM by default. (T or F)
Automatically configures the MGCP GW for you.
TRUE
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
41. What are the commands to create vlans on an ESW?
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
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.
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
42. What should you configure before entering auto qos?
Bottom up - to prevent glare.
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Bandwidth
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
43. A <i>show isdn status</i> shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
#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
8kb/s
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
(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
44. SNR is also known as?
Mobile Connect
...
(config)#voice register dialplan
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
45. 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.
(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
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
#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
46. When do you do digit manipulation at the route pattern as opposed to the route list?
To notify SIP Phones of NTP
...
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
47. When do you use translate called? Translate calling?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
#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
8kb/s
48. Where do you use VIA zone?
...
8kb/s
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
Mobile Voice Access
49. What is call-forward pattern used for?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Call forwarding between voip to voip (when CUBE is in play)
When configuring TEHO.
50. <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 PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
...
Apply <i>voice-class codec</i>
1 ... this is not optional!