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Test your basic knowledge |
CCIE Voice Test
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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 do you allow the Calling Name to be sent to the PSTN on a router?
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Show ccm-manager
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
Configure it on the route group through the route list - then it will be local to the route list.
2. How do you block an external call from being transferred back out to the pstn by an internal user?
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
(config)#<i>sh cdp neigh detail
...
3. What is CSIM?
Call Simulator. You can use to validate path from router to the PSTN.
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
Bandwidth
'Service Parameters --> <i>Auto Answer Timer</i>
4. What terminology translates to SRST?
Dtmf-relay h245-alpha
Reroute when there is a WAN Outage.
Sends the Calling Name.
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
5. 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?
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
#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
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.
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
6. How do you set the inter-digit timeout for SIP phones in CME?
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Dial-peers
(config)#voice register dialplan
Single Number Reach (Mobile Connect)
7. What are the commands to create vlans on an ESW?
At the Line or Device level. -- Sometimes unpredictable performance of assigned to the device though.
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
Apply <i>voice-class codec</i>
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit
8. How do you configure SRST?
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
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
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
9. When using <i>drop-through-option</i> What is the max number of huntgroups?
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
1 ... this is not optional!
(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
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
10. 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>.
Trunk Port Method (3750 or ESW)||interface fa0/1|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10|switchport voice vlan 20|spanning-tree portfast
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
Service Parameter --> Block OffNet To OffNet Transfer --> <i>True</i>
11. Is the order of the MRGs in an MRGL significant?
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
On CUCM it's identical to adding an H323 GW.
24kb/s
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
12. On the RDP What is the CSS used for?
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
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
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)
Mobile Voice Access
13. How do you configure CUCM redundancy on an H323 gateway?
Outbound dial-peers
#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
Put them in a route list.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
14. What is Mobile Voice Access?
24kb/s
<i>sh ephone reg</i>|and|<i>debug voip dialpeer</i>
When I call in from off-net to a DID number in-net ... once authenticated - your calls appear to come from the desk phone.
(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
15. What is the Service Parameter to allow the phone that is <i>picking ip</i> to auto answer?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
Master/Slave relationship. CUCM controls it.
<i>#after-hours block pattern</i>
<i>Auto Call Pickup Enabled</i>
16. What are the commands to configure a MGCP Gateway? (router)
8kb/s
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Telephony-service|no auto-reg-ephone|max-ephones <NUM>|max-dn <NUM> no-reg|ip source-address <IP> port 2000|create cnf-files
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
17. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
Apply it to the ephone.
You need dial-peers and to allow h323 to h323 communication.||voice service voip| allow-connections h323 to h323
Service Parameters --> <i>Mobile Voice Access Number</i>
...
18. How do you setup AutoRegistration in CUCM?
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Apply it to the ephone.
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Single Number Reach (Mobile Connect)
19. What dtmf-relay type do you use for an H323 GW?
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
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
Dtmf-relay h245-alpha
Reroute when here is WAN congestion.
20. Describe how you configure SIP URI functionality.
<i>h225 timeout tcp establish 3</i> within a h323 voice class||#voice class h323 1|#h225 timeout tcp establish 3
...
<i>Auto Call Pickup Enabled</i>
21. How do you prioritize route groups?
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
Call Simulator. You can use to validate path from router to the PSTN.
Put them in a route list.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
22. How do you create a trunk on the router side?
#voice service voip| #no supplementary-service h225-notify cid-update
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
#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
<b>CAT IOS</b>|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10 --> !!!DATA VLAN!!!
23. What does <i>rule 1 /^.*(....$)/ /1/</i> do?
CUCM OS Administration Settings --> NTP Servers
Matches any length dialed number and truncates it to 4 digits.
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
The Called/Calling Transformations are superceded.
24. What must you do for a BACD script to work on a CME router?
Higher
Debug isdn q931
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
Put them in a route list.
25. How do you configure phone ports on a 3750?
Call forwarding between voip to voip (when CUBE is in play)
<b>Access Port Method</b>||interface range fa0/1 - 3|switchport voice vlan 20|switchport access vlan 10|spanning-tree portfast
24kb/s
26. 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?
8kb/s
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)
Sends every digit to CUCM in real-time with digit-analysis being handled by CUCM.
SERVICE PARAMETER>|| Block Offnet to Offnet Transfer - Set to True
27. How would you set the the timer for Auto Answer?
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28. How do you set up redundancy on outbound dial-peers on an H323 gateway?
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Max-dn|max-ephone|ip source-address
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
29. What are the base telephony-service commands for CME?
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
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST
Telephony service|ip source-address 10.10.110.2|max-dn <NUM>|max-ephone <NUM>|create cnf-files
...
30. An MGCP gateway serving as an SRST router requires what?
(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|(config-controller)#no shut||(config)#voice service voip|(conf-voi-serv)#sip|(conf-serv-sip)#bind
Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
System Cisco Unified CM|Uncheck the box <i>Auto-Reg Disabled</i>||Enterprise Parameters --> <i>Auto Registration Phone Protocol</i> SIP or SCCP
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
31. 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?
Automatically configures the MGCP GW for you.
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
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
Yes. The media resources in the first MRG listed will be used before media resources in the second MGR listed.
32. How do you prevent toll fraud on CME?
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
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 ....
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
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
33. If the PSTN is presenting on channel 1 - should you configure CUCM to search 'top down' or 'bottom up'?
Matches any length dialed number and truncates it to 4 digits.
Bottom up - to prevent glare.
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
Need to Know||Anytime you have an external call - it needs to match on a route pattern.
34. How do you prevent H323 caller-id updates to CUCM
Show ccm-manager
Configure it on the route group through the route list - then it will be local to the route list.
#voice service voip| #no supplementary-service h225-notify cid-update
<i>voice register pool</i>|and|<i>voice register dn</i>
35. How do you convert a full PRI 9config'd by ccm-manager config server) back to a Fractional PRI?
Single Number Reach (Mobile Connect)
Trunk Port Method (3750 or ESW)||interface fa0/1|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10|switchport voice vlan 20|spanning-tree portfast
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
36. How do you enable AAR?
Reroute when there is a WAN Outage.
...
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
SERVICE PARAMETER> Automated Alternate Routing Enable > True
37. How do you configure phone ports on an ESW?
#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
Trunk Port Method (3750 or ESW)||interface fa0/1|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10|switchport voice vlan 20|spanning-tree portfast
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1
Matches any length dialed number and truncates it to 4 digits.
38. Name 4 useful show commands for active calls.
#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
Apply <i>voice-class codec</i>
'Service Parameters --> <i>Auto Answer Timer</i>
#show call active brief|#show voice call summary|#show voice call status|show dialpeer voice summary
39. Maximum Wait Time for Desk Pickup?
Precede the # with a > ... so 9011*>#
<i>Proxy Line Mode</i> has multiple icons on the phone.|<i>Shared Line Mode</i> has only one.
In milliseconds ... allows you to hang up cell phone then pick up at the desk phone without disconnecting the call.
Trunk Port Method (3750 or ESW)||interface fa0/1|switchport trunk encap dot1q|switchport mode trunk|switchport trunk native vlan 10|switchport voice vlan 20|spanning-tree portfast
40. Calls coming from CUCM to PSTN need what?
(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|(config-controller)#no shut||(config)#voice service voip|(conf-voi-serv)#sip|(conf-serv-sip)#bind
Put them in a route list.
Outbound dial-peers
41. What are the 3 mandatory commands within call-manager-fallback?
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
To enable two-stage dialing.
SIP Dial Rules
Max-dn|max-ephone|ip source-address
42. What is SIP URI?
Phone number followed by domain name.|i.e. 3006@ipxcme.com
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
The Called/Calling Transformations are superceded.
(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
43. What are the commands to manually configure an MGCP gateway?
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
#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
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.
#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
44. How much bandwidth does a G.729 call including layer 3 require?
Sends the Calling Name.
24kb/s
Call forwarding between voip to voip (when CUBE is in play)
Dtmf-relay h245-alpha
45. If you're using Called/Calling Party Transformations at the Route Pattern - what happens if you do Digit manipulation at the route list?
'Service Parameters --> <i>Auto Answer Timer</i>
To enable two-stage dialing.
Sh ccm-manager
The Called/Calling Transformations are superceded.
46. '<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
Mobile Voice Access
...
Reroute when there is a WAN Outage.
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
47. What does Display-IE do?
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
In CUCM on the PRI interface of the MGCP Gateway - set the Significant Digits to 4.
Debug isdn q931
Sends the Calling Name.
48. What is a best practice for digit manipulation - in regards to H323 GWs?
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
Gatekeeper| security acl answerarq 1||set up an access list permit whoever you want to allow
CUCM OS Administration Settings --> NTP Servers
Push as much digit manipulation to the router as possible - so you can re-use it for SRST.
49. In a SIP Dial Rule how do you represent a dialed * instead of it being used as a wildcard?
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50. How do you configure a SIP Trunk? (router side)
(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|(config-controller)#no shut||(config)#voice service voip|(conf-voi-serv)#sip|(conf-serv-sip)#bind
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>
#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
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