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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 inform a SIP phone of NTP information?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
...
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
2. With the gw-priority command - does higher or lower priority take precedence?
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
Outbound dial-peers
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
Higher
3. Calls coming from CUCM to PSTN need what?
In CUCM - configure CFUR to point to it's E164 number.
Outbound dial-peers
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
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
4. When do you do digit manipulation at the route pattern as opposed to the route list?
Bottom up - to prevent glare.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
When multiple sites use the same route pattern - do your digit manipulation on a route list.
The Called/Calling Transformations are superceded.
5. Privacy is enabled system-wide in CUCM by default. (T or F)
TRUE
#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
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.
(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
6. What should you configure before entering auto qos?
<i>voice register pool</i>|and|<i>voice register dn</i>
Bandwidth
Sh ip rsvp reservation||sh sccp connections
7. How much bandwidth does a G.729 call including layer 3 require?
Single Number Reach (Mobile Connect)
24kb/s
The Called/Calling Transformations are superceded.
<i>SIP Route Pattern</i> over a SIP Trunk.
8. How do you verify where MOH is being served up from?
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9. How do you enable AAR system wide?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
#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 before the ... so XXXX
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
10. On an MGCP GW - how could you see the primary and backup CUCM servers?
Show ccm-manager
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
11. SNR is also known as?
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
#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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Mobile Connect
12. Maximum Wait Time for Desk Pickup?
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
13. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
<i>Auto Call Pickup Enabled</i>
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
14. How do you configure DHCP in IOS?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
At the CLI: <i>utils ntp status</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
15. What are the steps to configure Single Number Reach and Mobile Voice Access?
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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
16. Where can you assign the AAR Group?
Sends the Calling Name.
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Show ccm-manager
Phone number followed by domain name.|i.e. 3006@ipxcme.com
17. If you have an H323 Gateway at a siet where the phones are controlled by CUCM - what do you need from a dial-peer perspective?
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.
#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
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 .)
18. What are the commands to configure NTP in IOS?
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Debug ephone moh
#voice service voip| #no supplementary-service h225-notify cid-update
<i>#after-hours block pattern</i>
19. 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
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.
Via the <i>voice hunt-group parallel</i> command
20. How do you prioritize route groups?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Put them in a route list.
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 .)
21. What is Mobile Voice Access?
Apply <i>voice-class codec</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.
(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.
22. What are the commands to set up the PRI?
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23. When you make a call and it results in a secondary dial-tone - What is this a symptom of?
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
Single Number Reach (Mobile Connect)
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Configure it on the route group through the route list - then it will be local to the route list.
24. What are the commands to configure SRST in fallback?
SIP Dial Rules
Call Simulator. You can use to validate path from router to the PSTN.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
25. What is the bit rate for a G.729 call excluding layer 2?
8kb/s
The <i>name</i> configuration field in ephone-dn and voice register dn
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
26. When using <i>drop-through-option</i> What is the max number of huntgroups?
1 ... this is not optional!
(config)#<i>sh cdp neigh detail
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
To notify SIP Phones of NTP
27. On the RDP What is the Rerouting CSS used for?
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Single Number Reach (Mobile Connect)
(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
Call forwarding between voip to voip (when CUBE is in play)
28. How do you configure class of service (CoS) in CUCM? CME?
The Called/Calling Transformations are superceded.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Reroute when there is a WAN Outage.
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 ....
29. On the RDP What is the CSS used for?
(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
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Mobile Voice Access
30. What does Display-IE do?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
1 ... this is not optional!
Debug isdn q931
Sends the Calling Name.
31. '<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
When multiple sites use the same route pattern - do your digit manipulation on a route list.
...
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
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
32. What types of digit manipulation can you perform at the route pattern?
<i>Auto Call Pickup Enabled</i>
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
33. How do you ensure that G.711 only is used?
Put them in a route list.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
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
34. 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.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Mobile Connect
35. 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
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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Apply it to the ephone.
36. What kind of relationship does an MGCP gateway have with CUCM?
Master/Slave relationship. CUCM controls it.
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
<i>voice register pool</i>|and|<i>voice register dn</i>
37. When are digits stripped in a gateway?
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Assign a logout-profile to the ephone.
When they are explicitly matched in a destination-pattern in a dial-peer.
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
38. When can you not use a Standard Local Route Group?
(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
Assign a logout-profile to the ephone.
Ccm-manager music-on-hold
When configuring TEHO.
39. How do you configure SRST?
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 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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
40. How do you create a trunk on the switch side?
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
41. What are the 3 mandatory commands within call-manager-fallback?
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Higher
Max-dn|max-ephone|ip source-address
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
42. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?
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43. How do you disable KPML?
Higher
SERVICE PARAMETER> Automated Alternate Routing Enable > True
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Assign SIP Dial Rules
44. How do you configure AAR?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
At the CLI: <i>utils ntp status</i>
<i>#after-hours block pattern</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.
45. What are the commands to create vlans on an ESW?
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
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.
(config)#voice register dialplan
46. 'If you're not getting a DHCP address from CUCM what then?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
<i>Auto Call Pickup Enabled</i>
47. How do you allow the Calling Name to be sent to the PSTN on a router?
Debug isdn q931
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
48. How do you test a Voice Translation Rule?
Matches any length dialed number and truncates it to 4 digits.
#test voice translation rule 1 <input to test>
(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
49. What is the full E164 format?
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
On CUCM it's identical to adding an H323 GW.
#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
50. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
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
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.