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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 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
Allows you to transfer by only pressing the Transfer button once.
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. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
When configuring TEHO.
3. 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 before the ... so XXXX
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)
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
4. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
CUCM OS Administration Settings --> NTP Servers
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
5. Maximum Wait Time for Desk Pickup?
IP Voice Media Streaming App
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Device Pool Locations
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
6. Is the + character supported on a VOIP dial-peer?
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7. What are the 3 mandatory commands within call-manager-fallback?
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
Max-dn|max-ephone|ip source-address
Phone number followed by domain name.|i.e. 3006@ipxcme.com
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
8. Where can you find the command to restore any of the default IP Phone Services if they happen to have been deleted?
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
9. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
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
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Service Parameters --> <i>Mobile Voice Access Number</i>
10. What are the commands to manually configure an MGCP gateway?
<i>SIP Route Pattern</i> over a SIP Trunk.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
(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
#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
11. When using GK redundancy - what must you do to ensure both SUB and PUB use port 1720 for call-signalling.
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12. How do you configure SRST?
Call-manager-fallback</i> or <i>telephony-service mode SRST</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
'Service Parameters --> <i>Auto Answer Timer</i>
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
13. How do you set the inter-digit timeout for SIP phones in CME?
(config)#voice register dialplan
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Phone number followed by domain name.|i.e. 3006@ipxcme.com
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
14. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
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
Matches any length dialed number and truncates it to 4 digits.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
To enable two-stage dialing.
15. What are the commands to configure SIP phones in CME?
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
Service Parameters --> <i>Mobile Voice Access Number</i>
TRUE
(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
16. Two useful troubleshooting commands for CUCME?
When they are explicitly matched in a destination-pattern in a dial-peer.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
It's best to strip digits at the voice port.
17. When can you not use a Standard Local Route Group?
When multiple sites use the same route pattern - do your digit manipulation on a route list.
When configuring TEHO.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
18. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
The <i>name</i> configuration field in ephone-dn and voice register dn
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Ccm-manager music-on-hold
19. What types of digit manipulation can you perform at the route pattern?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Service Parameters --> <i>Mobile Voice Access Number</i>
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
20. What kind of relationship does an MGCP gateway have with CUCM?
Master/Slave relationship. CUCM controls it.
...
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
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
21. Name 4 useful show commands for active calls.
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
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.
Show ccm-manager
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
22. How is NTP sych setup in CUCM?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Call Simulator. You can use to validate path from router to the PSTN.
CUCM OS Administration Settings --> NTP Servers
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
23. How do you configure AAR?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
When they are explicitly matched in a destination-pattern in a dial-peer.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
24. When configuring a Gatekeep-controlled trunk - what do you do with the <i>Wait for Far End H.245 TCS</i> option?
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.
At the CLI: <i>utils ntp status</i>
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
25. What do you need to do to activate the CME GUI?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
#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
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
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
26. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
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.
The Called/Calling Transformations are superceded.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
27. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
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
#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
#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
<i>Auto Call Pickup Enabled</i>
28. An MGCP gateway serving as an SRST router requires what?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Outbound dial-peers
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Bandwidth
29. Where do you use VIA zone?
Call Simulator. You can use to validate path from router to the PSTN.
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
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Show ccm-manager
30. What should you configure before entering auto qos?
Bandwidth
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
<i>voice register pool</i>|and|<i>voice register dn</i>
31. What is the bit rate for a G.729 call excluding layer 2?
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
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
8kb/s
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
32. How do you verify that NTP is working on the CUCM server?
At the CLI: <i>utils ntp status</i>
To enable two-stage dialing.
Sh ip rsvp reservation||sh sccp connections
(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
33. What does <i>ccm-manager config server [IP]</ip> do?
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
Automatically configures the MGCP GW for you.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
Put them in a route list.
34. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
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
<i>#after-hours block pattern</i>
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
35. What's the best way to perform digit manipulation on a route group?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Automatically configures the MGCP GW for you.
Configure it on the route group through the route list - then it will be local to the route list.
<i>SIP Route Pattern</i> over a SIP Trunk.
36. If you're using Called/Calling Party Transformations at the Route Pattern and tranformations at the route list ... What does the RP affect and What does the RL affect?
#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
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>
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
37. What are the commands to create the L3 routing interface for VLANS (SVI)?
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.
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
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
38. What is Phone NTP Reference used for?
To notify SIP Phones of NTP
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Call Simulator. You can use to validate path from router to the PSTN.
...
39. Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Automatically configures the MGCP GW for you.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
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.
40. How would you set the the timer for Auto Answer?
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41. Is the order of the MRGs in an MRGL significant?
Device Pool Locations
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
<i>voice register pool</i>|and|<i>voice register dn</i>
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
42. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
On CUCM it's identical to adding an H323 GW.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
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.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
43. What is a best practice for digit manipulation - in regards to H323 GWs?
8kb/s
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.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
44. On an H323 GW - how do you adjust the timers for redundancy hunting?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
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
The Called/Calling Transformations are superceded.
45. How do you create a trunk on the router side?
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
Apply it to the ephone.
Assign a logout-profile to the ephone.
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
46. How do you use an ephone template?
Apply it to the ephone.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Assign SIP Dial Rules
CUCM OS Administration Settings --> NTP Servers
47. How do you enable Extension Mobility for a device in 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 ....
Assign a logout-profile to the ephone.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Automatically configures the MGCP GW for you.
48. How do do configure TEHO?
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.
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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
49. What are the commands to configure a MGCP Gateway? (router)
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
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
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
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
50. What are the commands to configure SRST in fallback?
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
Call forwarding between voip to voip (when CUBE is in play)