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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. Aside from the <i>Media Resources --> Mobile Voice Access</i> where else do you need to specify the the Mobile Voice Access DN?
Higher
Service Parameters --> <i>Mobile Voice Access Number</i>
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
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
2. On the RDP What is the CSS used for?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Apply <i>voice-class codec</i>
Configure it on the route group through the route list - then it will be local to the route list.
Mobile Voice Access
3. How do you test a Voice Translation Rule?
Dial-peers
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.
#test voice translation rule 1 <input to test>
8kb/s
4. Privacy is enabled system-wide in CUCM by default. (T or F)
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
TRUE
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
Configure it on the route group through the route list - then it will be local to the route list.
5. Describe how you configure SIP URI functionality.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
Reroute when here is WAN congestion.
6. What are the commands to configure SRST in fallback?
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.
To notify SIP Phones of NTP
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
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
7. What dtmf-relay type do you use for an H323 GW?
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Put them in a route list.
Dtmf-relay h245-alpha
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
8. 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
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
#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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
9. How do you configure SRST?
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Outbound dial-peers
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 ....
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
10. Does CUCM support RSVP natively?
To enable two-stage dialing.
Dial-peers
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
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
11. What are the commands to configure a MGCP Gateway? (router)
#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>|| Block Offnet to Offnet Transfer - Set to True
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
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
12. What are the basic ephone-dn and ephone commands?
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
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
1 ... this is not optional!
13. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
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14. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
The <i>name</i> configuration field in ephone-dn and voice register dn
15. How do you prioritize route groups?
...
TRUE
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Put them in a route list.
16. What terminology translates to AAR?
Assign SIP Dial Rules
Reroute when here is WAN congestion.
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Higher
17. What is Mobile Voice Access?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
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.
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
18. How would you enable security on a GK?
Gatekeeper|endpoint resource-threshold|endpoint max-calls h323id gk-trunk_2 1||The bandwidth command only allows restriction between zones
Via the <i>voice hunt-group parallel</i> command
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
Precede the # with a > ... so 9011*>#
19. How can you tell the difference between Proxy Line Mode and Shared Line Mode by just looking at the phone?
Dial-peers
Allows you to transfer by only pressing the Transfer button once.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
20. How do you set the inter-digit timeout for SIP phones in CME?
24kb/s
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
(config)#voice register dialplan
21. Where can you assign the AAR Group?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
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
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
22. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
<i>#after-hours block pattern</i>
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
<i>Auto Call Pickup Enabled</i>
23. What does KPML do?
Debug isdn q931
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
24. 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
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
Max-dn|max-ephone|ip source-address
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
25. SNR is also known as?
Mobile Connect
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
#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
Dial-peers
26. How do you see multicast packets being sent?
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
Debug ephone moh
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
27. What terminology translates to SRST?
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Ccm-manager music-on-hold
#test voice translation rule 1 <input to test>
Reroute when there is a WAN Outage.
28. How do you enable Extension Mobility for a device in CME?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
Assign a logout-profile to the ephone.
The Called/Calling Transformations are superceded.
29. 'How do you inform a SIP phone of NTP information?
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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
SIP Dial Rules
30. What are the steps to integrate CUPS with CUCM?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
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
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
TRUE
31. 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.
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
SERVICE PARAMETER> Automated Alternate Routing Enable > True
32. How do you allow H323 calls to be preserved should the primary H323 GW fail?
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
Device Pool Locations
#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
33. Maximum Wait Time for Desk Pickup?
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
'Use before the ... so XXXX
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Master/Slave relationship. CUCM controls it.
34. How can you confirm the MGCP GW is registered to CUCM - in IOS?
It's best to strip digits at the voice port.
#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
Sh ccm-manager
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
35. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
Sh ccm-manager
Reroute when there is a WAN Outage.
36. What are the steps to configure Single Number Reach and Mobile Voice Access?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
To enable two-stage dialing.
Top Down means that channel 1 will be the first channel used to place outgoing calls.
37. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
<i>voice register pool</i>|and|<i>voice register dn</i>
1 ... this is not optional!
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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
38. Two useful troubleshooting commands for CUCME?
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
(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
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
39. What are the commands to ensure an H323 GW always uses the Sub as primary and Pub as a backup?
Higher
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
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
#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
40. 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?
TRUE
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)
Debug ephone moh
Dtmf-relay h245-alpha
41. How do you block calls under 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
<i>#after-hours block pattern</i>
Assign SIP Dial Rules
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
42. How do you support multiple codecs on a dialpeer?
When they are explicitly matched in a destination-pattern in a dial-peer.
Ccm-manager music-on-hold
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Apply <i>voice-class codec</i>
43. 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
IP Voice Media Streaming App
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>
44. How do you get around relying on DNS for your CUCMs?
Debug ephone moh
Ccm-manager music-on-hold
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
45. 'If you want to make changes to any softkeys where do you do it?
The <i>name</i> configuration field in ephone-dn and voice register dn
Dial-peers
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
1 ... this is not optional!
46. What does <i>ccm-manager config server [IP]</ip> do?
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Automatically configures the MGCP GW for you.
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
47. What types of digit manipulation can you perform at the route pattern?
<i>Auto Call Pickup Enabled</i>
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
8kb/s
48. On the RDP What is the Rerouting CSS used for?
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Single Number Reach (Mobile Connect)
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
To notify SIP Phones of NTP
49. When are digits stripped in a gateway?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Debug ephone moh
When they are explicitly matched in a destination-pattern in a dial-peer.
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
50. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
To enable two-stage dialing.
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
(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
24kb/s