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Test your basic knowledge |
CCIE Voice Test
Start Test
Study First
Subjects
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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 would you enable security on a GK?
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
SERVICE PARAMETER> Automated Alternate Routing Enable > True
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
2. How do you ensure an H323 GW always uses the Sub as primary and Pub as a backup?
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 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
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
3. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
Reroute when there is a WAN Outage.
<i>voice register pool</i>|and|<i>voice register dn</i>
#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
Sh ccm-manager
4. With the gw-priority command - does higher or lower priority take precedence?
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!
Higher
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
5. Name 2 commands to verify RSVP functionality.
Sh ip rsvp reservation||sh sccp connections
Matches any length dialed number and truncates it to 4 digits.
Allows you to transfer by only pressing the Transfer button once.
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
6. What are the base telephony-service commands for CME?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
...
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 ....
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
7. What is SIP URI?
<i>Auto Call Pickup Enabled</i>
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Phone number followed by domain name.|i.e. 3006@ipxcme.com
8. How do you verify that NTP is working on the CUCM server?
At the CLI: <i>utils ntp status</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:(SUB IP)| #dtmf-relay h245-alpha||#dial-peer voice 30001 voip| #destination-pattern 3...| #voice-clas
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
9. How do you prevent H323 caller-id updates to CUCM
(config)#voice register dialplan
#voice service voip| #no supplementary-service h225-notify cid-update
Ccm-manager music-on-hold
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
10. What is the bit rate for a G.729 call excluding layer 2?
8kb/s
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
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
11. Aside from the <i>Media Resources --> Mobile Voice Access</i> where else do you need to specify the the Mobile Voice Access DN?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Service Parameters --> <i>Mobile Voice Access Number</i>
To enable two-stage dialing.
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
12. What kind of relationship does an MGCP gateway have with CUCM?
Automatically configures the MGCP GW for you.
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Master/Slave relationship. CUCM controls it.
13. What is Mobile Voice Access?
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
Single Number Reach (Mobile Connect)
To notify SIP Phones of NTP
14. How do you maintain 4-digit dialing to phones that are running in an SRST scenario (WAN outage)?
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15. What is CSIM?
The <i>name</i> configuration field in ephone-dn and voice register dn
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
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Call Simulator. You can use to validate path from router to the PSTN.
16. How do you prevent toll fraud on CME?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
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 ....
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
To enable two-stage dialing.
17. What are the steps to configure Single Number Reach and Mobile Voice Access?
Show ccm-manager
Phone number followed by domain name.|i.e. 3006@ipxcme.com
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
18. How do you use the # as a string terminator within a SIP Dial Rule?
Precede the # with a > ... so 9011*>#
Apply <i>voice-class codec</i>
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
#test voice translation rule 1 <input to test>
19. 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.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
#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
20. How do you disable KPML?
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
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
Assign SIP Dial Rules
Put them in a route list.
21. An MGCP gateway serving as an SRST router requires what?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
#voice service voip| #no supplementary-service h225-notify cid-update
Phone number followed by domain name.|i.e. 3006@ipxcme.com
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
22. How do you prevent toll fraud on CUCM?
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
Sends the Calling Name.
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
23. How do you configure phone ports on a 3750?
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
SERVICE PARAMETER> Automated Alternate Routing Enable > True
CUCM OS Administration Settings --> NTP Servers
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
24. How do you ensure no inter-digit timeout fro SIP phones within CUCM?
SIP Dial Rules
Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
Show ccm-manager
25. In CME Where is the Calling Name derived from?
The <i>name</i> configuration field in ephone-dn and voice register dn
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
The Called/Calling Transformations are superceded.
Device Pool Locations
26. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Dial-peers
Matches any length dialed number and truncates it to 4 digits.
<i>Auto Call Pickup Enabled</i>
27. 'If you want to make changes to any softkeys where do you do it?
Apply it to the ephone.
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Matches any length dialed number and truncates it to 4 digits.
(config)#voice register dialplan
28. What does an H323 GW require that MGCP GWs do not?
#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
Matches any length dialed number and truncates it to 4 digits.
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
Dial-peers
29. How do you create a trunk on the switch side?
SERVICE PARAMETER> Automated Alternate Routing Enable > True
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
It's best to strip digits at the voice port.
30. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
It's best to strip digits at the voice port.
When configuring TEHO.
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
31. What's the best way to do digit manipulation on an IOS gateway?
#test voice translation rule 1 <input to test>
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
32. On the RDP What is the CSS used for?
Mobile Voice Access
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)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
<i>Auto Call Pickup Enabled</i>
33. What are the commands to configure SIP phones in CME?
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
Sh ip rsvp reservation||sh sccp connections
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
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
34. Describe how you configure SIP URI functionality.
(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
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
35. What are the commands to configure a T1/E1 PRI?
Reroute when here is WAN congestion.
(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
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
36. How do you create a trunk on the router side?
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
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
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
When configuring TEHO.
37. When it comes to H323 gateways - what should you always change - unless explicitly told not to?
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
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
#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
Precede the # with a > ... so 9011*>#
38. <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 os truly unsurpassed. I have opened up a wealth of opportunities for myself and my family and
In CUCM - configure CFUR to point to it's E164 number.
#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
(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
...
39. How do you allow the Calling Name to be sent to the PSTN on a router?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Higher
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
40. How do you enable AAR system wide?
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 .)
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
41. What must you do for a BACD script to work on a CME router?
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
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.
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
42. What's the difference between AAR and SRST?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Device Pool Locations
8kb/s
Show ccm-manager
43. How do you configure SRST?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Allows you to transfer by only pressing the Transfer button once.
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
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
44. When do you do digit manipulation at the route pattern as opposed to the route list?
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.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
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 ....
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
45. What is a best practice for digit manipulation - in regards to H323 GWs?
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
(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
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
46. How do you see the details of calls coming in and out of the PRI?
Debug isdn q931
On CUCM it's identical to adding an H323 GW.
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
47. When can you not use a Standard Local Route Group?
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
(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
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
When configuring TEHO.
48. What CM Service needs to be start in Serviceability for MOH to work?
<i>voice register pool</i>|and|<i>voice register dn</i>
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
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!
49. How do you block an external call from being transferred back out to the pstn by an internal user?
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
(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
(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
50. A <i>show isdn status</i> shows TEI_ASSIGNED. What do you do to get MULTIPLE_FRAME_ESTABLISHED?
Outbound dial-peers
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True