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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 setup AutoRegistration in CUCM?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
To notify SIP Phones of NTP
<i>#after-hours block pattern</i>
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
2. 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>
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Put them in a route list.
Sh ccm-manager
3. An MGCP gateway serving as an SRST router requires what?
#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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
4. How would you set the the timer for Auto Answer?
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5. How do you allow H323 calls to be preserved should the primary H323 GW fail?
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
On CUCM it's identical to adding an H323 GW.
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
6. What are the basic SCCP Commands fro telephony-service in CUCME?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
Dtmf-relay h245-alpha
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
#test voice translation rule 1 <input to test>
7. 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>.
(config)#voice register dialplan
The Called/Calling Transformations are superceded.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
8. '<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
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
...
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
9. 'If you want to make changes to any softkeys where do you do it?
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
When configuring TEHO.
Dial-peers
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
10. Maximum Wait Time for Desk Pickup?
To notify SIP Phones of NTP
(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
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
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.
11. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
Higher
Show ccm-manager
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
12. How do you support multiple codecs on a dialpeer?
Apply <i>voice-class codec</i>
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
13. How do you configure class of service (CoS) in CUCM? CME?
Dial-peers
Mobile Connect
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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
14. How do you prevent H323 caller-id updates to CUCM
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
#voice service voip| #no supplementary-service h225-notify cid-update
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
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
15. On an MGCP GW - how could you see the primary and backup CUCM servers?
Show ccm-manager
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
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 Voice Access
16. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
Configure it on the route group through the route list - then it will be local to the route list.
Call Simulator. You can use to validate path from router to the PSTN.
Device Pool Locations
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
17. How do you block an external call from being transferred back out to the pstn by an internal user?
#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
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
Device Pool Locations
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
18. What is Phone NTP Reference used for?
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
To notify SIP Phones of NTP
19. 'If you're not getting a DHCP address from CUCM what then?
8kb/s
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
20. What does <i>ccm-manager config server [IP]</ip> do?
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
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)
Automatically configures the MGCP GW for you.
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
21. How do you enable AAR?
SIP Dial Rules
...
Mobile Connect
SERVICE PARAMETER> Automated Alternate Routing Enable > True
22. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
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23. When can you not use a Standard Local Route Group?
#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
Max-dn|max-ephone|ip source-address
Outbound dial-peers
When configuring TEHO.
24. 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?
On CUCM it's identical to adding an H323 GW.
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
25. What are the basic ephone-dn and ephone commands?
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
#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
Single Number Reach (Mobile Connect)
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
26. How do you see the details of calls coming in and out of the PRI?
Debug isdn q931
Sh ccm-manager
(config)#voice register dialplan
Apply it to the ephone.
27. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
#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
<i>SIP Route Pattern</i> over a SIP Trunk.
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
28. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
(config)#<i>sh cdp neigh detail
Sends the Calling Name.
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>Auto Call Pickup Enabled</i>
29. How do you configure DHCP in IOS?
(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
<i>Auto Call Pickup Enabled</i>
Top Down means that channel 1 will be the first channel used to place outgoing calls.
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
30. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
Call Simulator. You can use to validate path from router to the PSTN.
Debug isdn q931
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.
Single Number Reach (Mobile Connect)
31. Is the + character supported on a VOIP dial-peer?
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32. Aside from the <i>Media Resources --> Mobile Voice Access</i> where else do you need to specify the the Mobile Voice Access DN?
'Use before the ... so XXXX
Dtmf-relay h245-alpha
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
Service Parameters --> <i>Mobile Voice Access Number</i>
33. <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
...
Ccm-manager music-on-hold
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
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
34. How do you block calls under call-manager-fallback?
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
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
<i>#after-hours block pattern</i>
8kb/s
35. How do you configure CUCM redundancy on an H323 gateway?
#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
Redundant dial plan. One on CUCM and one on the routers via 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
In CUCM - configure CFUR to point to it's E164 number.
36. On an H323 GW - how do you adjust the timers for redundancy hunting?
To notify SIP Phones of NTP
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
At the CLI: <i>utils ntp status</i>
It's best to strip digits at the voice port.
37. What is Mobile Voice Access?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
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
Outbound dial-peers
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
38. SNR is also known as?
#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
Sends the Calling Name.
Mobile Connect
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
39. What are the base telephony-service commands for CME?
(config)#voice register dialplan
Bottom up - to prevent glare.
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
<i>SIP Route Pattern</i> over a SIP Trunk.
40. 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
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
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
8kb/s
41. 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?
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.
Call forwarding between voip to voip (when CUBE is in play)
Bottom up - to prevent glare.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
42. 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>voice translation-profile</i>|and|<i>voice translation-rule</i>
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
43. How do you prevent toll fraud on CME?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
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 ....
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
TRUE
44. How do you create a trunk on the router side?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
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
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
At the CLI: <i>utils ntp status</i>
45. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
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>voice register pool</i>|and|<i>voice register dn</i>
Top Down means that channel 1 will be the first channel used to place outgoing calls.
46. In CME Where is the Calling Name derived from?
The <i>name</i> configuration field in ephone-dn and voice register dn
Reroute when here is WAN congestion.
Show ccm-manager
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
47. How do you set the inter-digit timeout for SIP phones in CME?
#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
Service Parameters --> <i>Mobile Voice Access Number</i>
(config)#voice register dialplan
48. What CUCM services should you activate?
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
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
49. On the RDP What is the Rerouting CSS used for?
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Single Number Reach (Mobile Connect)
24kb/s
50. 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?
(config)#<i>sh cdp neigh detail
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
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.