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• Answer 50 questions in 15 minutes.
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• Match each statement with the correct term.
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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 an H323 GW require that MGCP GWs do not?
Dial-peers
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
#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
Debug ephone moh


2. How do you verify that NTP is working on the CUCM server?
#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
At the CLI: <i>utils ntp status</i>
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
Need to Know||Anytime you have an external call - it needs to match on a route pattern.


3. How would you set the the timer for Auto Answer?

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4. 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?
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
#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
Apply it to the ephone.
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)


5. 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 Parameters --> <i>Auto Answer Timer</i>
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
8kb/s


6. How do do configure TEHO?
#voice service voip| #no supplementary-service h225-notify cid-update
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
Enable MOH on the CME.||#<i>telephony service</i>|#<i>moh music-on-hold.au</i>
...


7. How much bandwidth does a G.729 call including layer 3 require?
CUCM Service Parameters --> <i>Call Park Reversion Timer</i>
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs
24kb/s
Gw-priority command||#gatekeeper| #zone prefix CUCM 2... gw-priority 10 US_Trunk_2| #zone prefix CUCM 2... gw-priority 5 US_Trunk_1


8. What does the <i>Transfer On-hook Enabled</i> Service Parameter do?
Higher
Assign a logout-profile to the ephone.
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
Allows you to transfer by only pressing the Transfer button once.


9. How do you allow H323 calls to be preserved should the primary H323 GW fail?
Ccm-manager music-on-hold
Bandwidth
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
SERVICE PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False


10. What commands are needed to configure the voice register pool in CME?
Bottom up - to prevent glare.
CUCM - Partitions and Calling Search Spaces||CME - COR Lists
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
The <i>name</i> configuration field in ephone-dn and voice register dn


11. <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
SERVICE PARAMETER> Automated Alternate Routing Enable > True
...
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 PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False


12. On an MGCP GW - if you select <i>Top Down</i> under Channel Selection Order - What does that mean?
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
Top Down means that channel 1 will be the first channel used to place outgoing calls.
<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).


13. An MGCP gateway serving as an SRST router requires what?
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 .)
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
(config)#voice register dialplan
Allows you to transfer by only pressing the Transfer button once.


14. What's the best way to do digit manipulation on an IOS gateway?
You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
8kb/s
<i>SIP Route Pattern</i> over a SIP Trunk.


15. How do you use an ephone template?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Apply it to the ephone.
Via the <i>voice hunt-group parallel</i> command
SIP Dial Rules


16. What does Display-IE do?
Sends the Calling Name.
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Use IPs instead of hostnames for all CUCM servers.|ENTERPRISE PARAMETERS --> Replace all CUCM hostnames with IPs in URLs


17. Which takes precedence Device Locations or Device Pool Locations?
When they are explicitly matched in a destination-pattern in a dial-peer.
Ccm-manager music-on-hold
Device Pool Locations
Redundant dial plan. One on CUCM and one on the routers via dial-peers.


18. On the RDP What is the Rerouting CSS used for?

Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
TRUE
Single Number Reach (Mobile Connect)


19. What command is needed on a gateway for it to invoke the DSP to convert multicast MOH stream to TDM?
Bottom up - to prevent glare.
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.
(config)#<i>sh cdp neigh detail
Ccm-manager music-on-hold


20. What is the full E164 format?
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
CM SERVICEABILITY TOOL --> SERVICE ACTIVATION --> Check all - then uncheck CMI and Security Services and activate.
Apply <i>voice-class codec</i>


21. How do you configure SIP phones to be able to call other SIP phones via SIP URI?
Assign SIP Dial Rules
#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 PARAMETER>|| Stop routing on unallocated number flag - Set to False|| Stop routing user busy - Set to False
<i>SIP Route Pattern</i> over a SIP Trunk.


22. When using <i>drop-through-option</i> What is the max number of huntgroups?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
1 ... this is not optional!


23. On an MGCP GW - how could you see the primary and backup CUCM servers?
Show ccm-manager
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
Put them in a route list.
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 .)


24. How do you configure AAR?
Debug isdn q931
1. Enable it.|2. Create AAR Group|3. Create CSS-AAR and apply at device level.|4. Apply AAR Group at line level.
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)


25. How do you see the details of calls coming in and out of the PRI?
Debug isdn q931
Mobile Voice Access
<i>voice translation-profile</i>|and|<i>voice translation-rule</i>
<i>voice register pool</i>|and|<i>voice register dn</i>


26. 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
Mobile Connect
+ [CC 1-3 digits] [National Number max 15 minus CC digits]
#test voice translation rule 1 <input to test>


27. How do you configure CUBE?

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28. With the gw-priority command - does higher or lower priority take precedence?
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.
Automatically configures the MGCP GW for you.
Higher
...


29. What terminology translates to SRST?

Region setting alone is not enough. You must set:|Enterprise Parameters --> Advertise G.722 Codec <i>Disable</i>
Debug ephone moh
Reroute when there is a WAN Outage.


30. How do you prioritize route groups?
Put them in a route list.
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 .)
No POTS dial-peer or no direct-inward-dial on a pots dial-peer
On CUCM it's identical to adding an H323 GW.


31. In CUCME what SIP config command is equivalent to ephone and ephone-dn?
<i>voice register pool</i>|and|<i>voice register dn</i>
...
Device Settings --> Softkey Template|Copy a template and make your changes.|Then apply to the device.
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.


32. Does CUCM support RSVP natively?
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.
Re-enter the <i>mgcp bind control</i> and <i>mgcp bind media</i> commands.
Assign a logout-profile to the ephone.
No it invokes the RSVP call agent on the IOS Router via SCCP message. i.e. use a Software MTP.


33. How do you verify where MOH is being served up from?

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34. When using GK redundancy - what must you do to ensure both SUB and PUB use port 1720 for call-signalling.

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35. 'How do you inform a SIP phone of NTP information?
System --> Phone NTP Reference|Then Apply to System --> Date/Time Group
The Called/Calling Transformations are superceded.
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 .)
Default is SRST||<i>mode cme</i> changes to CME|<i>no mode cme</i> changes back to SRST


36. How do you configure SRST?
Put them in a route list.
Call-manager-fallback</i> or <i>telephony-service mode SRST</i>
Max-dn|max-ephone|ip source-address
Uncheck it! Otherwise both H323 devices on either end wait for the other to send TCS first - and the connection will timeout.


37. What is the <i>Enable Enterprise Feature Access</i> Service Parameter used for?
DDI - Digit Discard Instruction (PreDot)||Mask - most powerful (does DDI and Prefix)||Prefix - Prefixes digits|| **These three types are compounded in this order!
#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
To enable two-stage dialing.
IP Voice Media Streaming App


38. How do you allow the Calling Name to be sent to the PSTN on a router?
<i>Auto Call Pickup Enabled</i>
(config)#int s0/0/0:15 (this is created after you setup the PRI)|(config-if)#isdn outgoing display-ie
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
Device Pool Locations


39. On the RDP What is the CSS used for?
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
Mobile Voice Access
To notify SIP Phones of NTP
Phone number followed by domain name.|i.e. 3006@ipxcme.com


40. 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
(config)#clock timezone|(config)#ntp server/master|(config)#sh ntp status to verify
Outbound dial-peers
Vlan database|vlan 130 name DATA|vlan 240 name PHONES|apply|exit


41. How do you enable AAR system wide?
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.
CUCM Service Parameters --> Automated Alternate Routing Enable --> <i>True</i>
#voice-class h323 1|#h225 timeout tcp establish 3 (if no response within 3 seconds continue dial-peer hunting)
#call-manager-fallback|#ip source-address (IP)|#max-ephone 4|max-dn 8|Configure a refernce in CUCM


42. What would force you to use telephony-service to configure SRST?
Redundant dial plan. One on CUCM and one on the routers via dial-peers.
Translate called - Calls going out to the PSTN||Translate Calling - Use on calls coming in from the PSTN. To change the ANI (Caller ID)
1. Complicated Hunt Group|2. Need to support Calling Name|3. Need to support Call Park Call Pickup
+ [CC 1-3 digits] [National Number max 15 minus CC digits]


43. What are the steps to configure Single Number Reach and Mobile Voice Access?
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
+ [CC 1-3 digits] [National Number max 15 minus CC digits]

You may be running into a CSA bug. Disable CSA in CUCM by:|SSH to CUCM|<i>utils csa disable</i>


44. What is call-forward pattern used for?
Call forwarding between voip to voip (when CUBE is in play)

(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
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.


45. When a device is roaming to a different site How does it get the <i>Roaming Sensitive Settings</i> from the Device Pool?
Bottom up - to prevent glare.
When multiple sites use the same route pattern - do your digit manipulation on a route list.
Bandwidth
1st the Subnet is recognized. (Device Mobility Info)|Then different Physical Locations are compared.


46. If you need terminate and reoriginate all calls through the CUBE - what do you need to do.
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
<i>voice register pool</i>|and|<i>voice register dn</i>
24kb/s


47. Describe how you configure SIP URI functionality.

In the CUCM 7.0.1 Release Notes ... search for <i>restore</i>.
Ssh to the CUCM the phone is registered to|show perf query class 'Cisco MOH Device'
On CUCM it's identical to adding an H323 GW.


48. Is the + character supported on a VOIP dial-peer?

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49. How do you configure class of service (CoS) in CUCM? CME?
CUCM - Partitions and Calling Search Spaces||CME - COR Lists

Enterprise Parameters --> <i>Organization Top Level Domain</i>|Enterprise Parameters --> <i>Cluster Fully Qualified Domain Name</i>
1 ... this is not optional!


50. What is the bit rate for a G.729 call excluding layer 2?
AAR -- Used for WAN Congestion|SRST -- Used for WAN Outage
8kb/s
RP affects the display on the calling phone.||RL affects What is sent to the gateway (called number).
Assign a logout-profile to the ephone.