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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. A <i>show isdn status</i> shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
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 .)
Outbound dial-peers
Debug isdn q931
2. On the RDP What is the Rerouting CSS used for?
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
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]
Single Number Reach (Mobile Connect)
3. How do you test a Voice Translation Rule?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
#test voice translation rule 1 <input to test>
4. How do you support multiple codecs on a dialpeer?
(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
(config)#<i>sh cdp neigh detail
Apply <i>voice-class codec</i>
Phone number followed by domain name.|i.e. 3006@ipxcme.com
5. How do you place SCCP and SIP phones into a single huntgroup?
When they are explicitly matched in a destination-pattern in a dial-peer.
Via the <i>voice hunt-group parallel</i> command
Ccm-manager music-on-hold
Automatically configures the MGCP GW for you.
6. How do you block an external call from being transferred back out to the pstn by an internal user?
Call Simulator. You can use to validate path from router to the PSTN.
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.
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
(config)#voice register dialplan
7. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?
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8. How do you prevent H323 caller-id updates to CUCM
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
Show ccm-manager
#voice service voip| #no supplementary-service h225-notify cid-update
When they are explicitly matched in a destination-pattern in a dial-peer.
9. How do you verify where MOH is being served up from?
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10. In Gatekeeper CAC how do you restrict a specific endpoint?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Reroute when there is a WAN Outage.
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
11. Maximum Wait Time for Desk Pickup?
(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
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
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
12. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
Reroute when there is a WAN Outage.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
At the CLI: <i>utils ntp status</i>
13. An MGCP gateway serving as an SRST router requires what?
Sends the Calling Name.
In CUCM - configure CFUR to point to it's E164 number.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
14. When configuring a Gatekeep-controlled trunk - what do you do with the <i>Wait for Far End H.245 TCS</i> option?
Device Pool Locations
...
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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
15. How do you set the inter-digit timeout for SIP phones in CME?
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.
Sh ip rsvp reservation||sh sccp connections
(config)#voice register dialplan
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
16. What is a best practice for digit manipulation - in regards to H323 GWs?
Master/Slave relationship. CUCM controls it.
Outbound dial-peers
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
(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
17. 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
SIP Dial Rules
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
18. What are the commands to create the L3 routing interface for VLANS (SVI)?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
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 .)
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
Configure it on the route group through the route list - then it will be local to the route list.
19. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
Matches any length dialed number and truncates it to 4 digits.
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Show ccm-manager
20. How do you configure CUCM redundancy on an H323 gateway?
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
#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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
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.
21. 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)
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
On CUCM it's identical to adding an H323 GW.
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
22. On an H323 GW - how do you adjust the timers for redundancy hunting?
<i>voice register pool</i>|and|<i>voice register dn</i>
SIP Dial Rules
Sh ccm-manager
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
23. Where is it best to manipulate digits for inbound calls?
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24. What are the commands to set up the PRI?
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25. How do you disable KPML?
#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
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Assign SIP Dial Rules
Top Down means that channel 1 will be the first channel used to place outgoing calls.
26. What are the commands to configure a SIP phone in CUCME?
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
8kb/s
In CUCM - configure CFUR to point to it's E164 number.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
27. How do you enable AAR system wide?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
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.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
28. What are the commands to configure NTP in IOS?
Bandwidth
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Higher
Single Number Reach (Mobile Connect)
29. What are the commands to configure an H323 GW?
#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
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
Show ccm-manager
When configuring TEHO.
30. Is the + character supported on a VOIP dial-peer?
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31. 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.
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
32. On the RDP What is the CSS used for?
Mobile Voice Access
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
33. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
#test voice translation rule 1 <input to test>
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.
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
34. How do you configure DHCP in IOS?
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
(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
(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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
35. What is the bit rate for a G.729 call excluding layer 2?
TRUE
<i>#after-hours block pattern</i>
8kb/s
36. What does Display-IE do?
Sends the Calling Name.
#test voice translation rule 1 <input to test>
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Call forwarding between voip to voip (when CUBE is in play)
37. What is call-forward pattern used for?
Call forwarding between voip to voip (when CUBE is in play)
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 Connect
38. How do you ensure that G.711 only is used?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
39. How do do configure TEHO?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
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
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
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
40. When dealing with an MGCP GW there is no intelligence on the router. How would you strip an incoming Called Number from 10 to 4 digits?
...
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
...
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
41. Which takes precedence Device Locations or Device Pool Locations?
Device Pool Locations
CUCM OS Administration Settings --> NTP Servers
'Service Parameters --> <i>Auto Answer Timer</i>
42. How do you get around relying on DNS for your CUCMs?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Apply it to the ephone.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Single Number Reach (Mobile Connect)
43. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
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
Put them in a route list.
<i>SIP Route Pattern</i> over a SIP Trunk.
8kb/s
44. How do you configure an MGCP GW? (router side)
(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
In CUCM - configure CFUR to point to it's E164 number.
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
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.
45. What does the <i>Transfer On-hook Enabled</i> Service Parameter do?
Allows you to transfer by only pressing the Transfer button once.
(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
(config)#voice register dialplan
At the CLI: <i>utils ntp status</i>
46. 'How do you inform a SIP phone of NTP information?
IP Voice Media Streaming App
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
47. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
SIP Dial Rules
The <i>name</i> configuration field in ephone-dn and voice register dn
#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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
48. What are the commands to manually configure an MGCP gateway?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
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
#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
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
49. 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?
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)
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.
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
50. On a gatekeeper trunk - how do you ensure the Sub is used as primary and Pub as backup?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Via the <i>voice hunt-group parallel</i> command
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
To notify SIP Phones of NTP