Application Note for Configuring Avaya Aura Manager R6.2 and … · 2014-10-02 · Manager R6.2 and Avaya Aura® Communication Manager R6.0.1 with Cisco 7941G, Cisco7942G and Cisco
This document is posted to help you gain knowledge. Please leave a comment to let me know what you think about it! Share it to your friends and learn new things together.
Transcript
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
Table of Contents Table of Contents ............................................................................................................................ 2
2.1. Verify Avaya Aura® Communication Manager License
User the display system-parameter customer options command to compare the Maximum
Administered SIP Trunks field value with the corresponding value in the USED column. The
difference between the two values needs to be greater than or equal to the desired number of
simultaneous SIP trunk connections.
Note: The license file installed on the system controls the maximum features permitted. If there
is insufficient capacity or a required feature is not enabled, contact an authorized Avaya sales
representative to make the appropriate changes.
display system-parameters customer-options Page 2 of 11 OPTIONAL FEATURES IP PORT CAPACITIES USED Maximum Administered H.323 Trunks: 12000 0 Maximum Concurrently Registered IP Stations: 18000 1 Maximum Administered Remote Office Trunks: 12000 0 Maximum Concurrently Registered Remote Office Stations: 18000 0 Maximum Concurrently Registered IP eCons: 414 0 Max Concur Registered Unauthenticated H.323 Stations: 100 0 Maximum Video Capable Stations: 18000 0 Maximum Video Capable IP Softphones: 18000 0 Maximum Administered SIP Trunks: 24000 10 Maximum Administered Ad-hoc Video Conferencing Ports: 24000 0 Maximum Number of DS1 Boards with Echo Cancellation: 522 0 Maximum TN2501 VAL Boards: 128 0 Maximum Media Gateway VAL Sources: 250 0 Maximum TN2602 Boards with 80 VoIP Channels: 128 0 Maximum TN2602 Boards with 320 VoIP Channels: 128 1 Maximum Number of Expanded Meet-me Conference Ports: 300 0
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
Use the change system-parameters features command to allow for trunk-to-trunk transfers.
This feature is needed to allow for transferring an incoming/outgoing call from /to a remote
switch back out to the same or different switch. For simplicity, the Trunk-to-Trunk Transfer
field was set to all to enable trunk-to-trunk transfer on a system wide basis.
change system-parameters features Page 1 of 19 FEATURE-RELATED SYSTEM PARAMETERS Self Station Display Enabled? y Trunk-to-Trunk Transfer: all Automatic Callback with Called Party Queuing? n Automatic Callback - No Answer Timeout Interval (rings): 3 Call Park Timeout Interval (minutes): 1 Off-Premises Tone Detect Timeout Interval (seconds): 20 AAR/ARS Dial Tone Required? y Music (or Silence) on Transferred Trunk Calls? no DID/Tie/ISDN/SIP Intercept Treatment: attd Internal Auto-Answer of Attd-Extended/Transferred Calls: transferred Automatic Circuit Assurance (ACA) Enabled? n
2.3. Administer IP Node Names
Use the change node-names-ip command to add entries for the Communication Manager and
Session Manager that will be used for connectivity. In the sample network clan and
192.168.81.104 are entered as Name and IP Address for the CLAN card in Communication
Manager running on the Avaya S8800 Server. In addition, sm62vl81 and 192.168.81.119 are
entered for the Session Manager. (The identify sm62vl81 is the hostname of Session Manager
server.)
change node-names ip Page 1 of 2 IP NODE NAMES Name IP Address clan 192.168.81.104 default 0.0.0.0 gateway 192.168.81.254 medpro 192.168.81.105 procr 192.168.81.102 procr6 :: sm62vl81 192.168.81.119
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
In the test configuration, Communications Manager acts an Evolution Server. An IMS enabled
SIP trunk is not required. The example uses signal group 150 in conjunction with Trunk Group
150 to reach the Session Manager. Use the add signalling-group n command where n is the
signalling group number being added to the system. Use the values defined in Sections 2.3 and
2.4 for the Near-end Node name, Far-end Node name and Far-end Network Region. The
Far-end Domain is left blank so that the signalling accepts any authoritative domain or have a
domain entered if preferred. Set IMS enabled to n and Peer Detection Enabled to y. Set Direct
IP-IP Audio Connections to y to turn “shuffling” on.
add signaling-group 150 Page 1 of 1 SIGNALING GROUP Group Number: 150 Group Type: sip IMS Enabled? n Transport Method: tcp Q-SIP? n SIP Enabled LSP? n IP Video? n Enforce SIPS URI for SRTP? y Peer Detection Enabled? y Peer Server: SM Near-end Node Name: clan Far-end Node Name: sm62vl81 Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: 1 Far-end Domain:mmsil.local Bypass If IP Threshold Exceeded? n Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y Session Establishment Timer(min): 3 IP Audio Hairpinning? n Enable Layer 3 Test? y Initial IP-IP Direct Media? n H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
To add a corresponding trunk group use the command add trunk-group n, where n is the trunk
group number.
• Group Number Set from the add-trunk-group n command
• Group Type Set as sip
• COR Set Class of Restriction (default 1)
• TN Set Tenant Number (default 1)
• TAC Choose integer value usually set the same as Trunk Group
number
• Direction Set to two-way
• Group Name Choose an appropriate name
• Outgoing Display Set to y
• Service Type Set to tie
• Signaling Group Enter the corresponding Signaling group number
• Number of Members Enter the number of members (trunk lines will automatically
assign when form is submitted.)
add trunk-group 150 Page 1 of 21 TRUNK GROUP Group Number: 150 Group Type: sip CDR Reports: y Group Name: SIP TG COR: 1 TN: 1 TAC: 150 Direction: two-way Outgoing Display? y Dial Access? n Night Service: Queue Length: 0 Service Type: tie Auth Code? n Member Assignment Method: auto Signaling Group: 150 Number of Members: 10
Navigate to Page 3 and set Numbering Format to private.
add trunk-group 150 Page 3 of 21 TRUNK FEATURES ACA Assignment? n Measured: none Maintenance Tests? y Numbering Format: private UUI Treatment: service-provider Replace Restricted Numbers? n Replace Unavailable Numbers? n Modify Tandem Calling Number: no Show ANSWERED BY on Display? y
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
Navigate to Page 4 and enter 101 for the Telephone Event Payload Type and P-Asserted-
Identity for Identity for Calling Party Display.
display trunk-group 150 Page 4 of 21 PROTOCOL VARIATIONS Mark Users as Phone? n Prepend '+' to Calling Number? n Send Transferring Party Information? y Network Call Redirection? n Send Diversion Header? n Support Request History? y Telephone Event Payload Type: 101 Convert 180 to 183 for Early Media? n Always Use re-INVITE for Display Updates? n Identity for Calling Party Display: P-Asserted-Identity Enable Q-SIP? n
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
Configure a route pattern to correspond to the newly added SIP trunk group. Use the change
route-pattern n command, where n is the route pattern number. Commonly this may match the
trunk group number for consistency of programming. Configure this route pattern to route calls
to trunk group 150, as configured in Section 2.5.2. Assign the lowest FRL (facility restriction
level) to allow all callers to use this route pattern, Assign 0 to No. Del Digits.
change route-pattern 150 Page 1 of 3 Pattern Number: 150 Pattern Name: To SessMan SCCAN? n Secure SIP? n Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC No Mrk Lmt List Del Digits QSIG Dgts Intw 1: 150 0 0 n user 2: n user 3: n user 4: n user 5: n user 6: n user BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR 0 1 2 M 4 W Request Dgts Format Subaddress 1: y y y y y n n unre none 2: y y y y y n n rest none 3: y y y y y n n rest none 4: y y y y y n n rest none 5: y y y y y n n rest none 6: y y y y y n n rest none
2.7. Administer Private Numbering
Use the change private-numbering command to define the calling part number to be sent out
through the SIP trunk. In the sample network configuration, all calls originating from a 5 digit
extension beginning with 24 will result in a 5-digit calling number. The calling party number will
be in the SIP “From” header.
change private-numbering 0 Page 1 of 2 NUMBERING - PRIVATE FORMAT Ext Ext Trk Private Total Len Code Grp(s) Prefix Len 5 23 150 5 Total Administered: 9 5 24 150 5 Maximum Entries: 540 5 37 150 5 5 38000 199 5 5 38001 199 5 5 38002 199 5 5 38111 150 5 5 38222 150 5 5 38888 150 5
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
Use the change locations to define the proxy route to use for outgoing calls. In the sample
network, the proxy route will be the trunk group defined in Section 2.5.2.
change locations Page 1 of 1 LOCATIONS ARS Prefix 1 Required For 10-Digit NANP Calls? y Loc Name Timezone Rule NPA Proxy Sel No Offset Rte Pat 1: Main + 00:00 0 150
2.9. Administer Dial Plan and AAR Analysis
Configure the dial plan for dialling 5-digit extension patterns beginning with 24 to SIP stations
registered with the Avaya. Use the change dialplan analysis command to define Dialed String
24 as an ext Call Type.
Change dialplan analysis Page 1 of 12 DIAL PLAN ANALYSIS TABLE Location: all Percent Full: 4 Dialed Total Call Dialed Total Call Dialed Total Call String Length Type String Length Type String Length Type 1 3 dac 38001 5 aar 2 5 aar 38002 5 aar 20 4 aar 38111 5 aar 230 5 ext 38222 5 aar 231 5 ext 38888 5 aar 232 5 ext 50 4 aar 233 5 ext 555 5 aar 235 5 ext 799 3 fac 23998 5 aar 81 6 aar 23999 5 aar * 3 fac 24 5 ext # 3 fac 25 4 aar 35 5 aar 37 5 aar 38000 5 aar
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
Under Name enter a suitable identifier. Under Description enter a suitable description. From the
Available Applications section, select the + sign beside the Application that is to be added to this sequence. Verify that the Application in this Sequence is updated correctly Click Commit
to save.
At this point the configuration of ASM is complete. To add users for Avaya SIP endpoints refer
to Section 3.12.
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
<processNodeName>192.168.81.119</processNodeName> --identifies the Avaya SIP Proxy </callManager> </member> </members> </callManagerGroup> <loadInformation124 model="Cisco IP Phone 7914 14-Button Line Expansion Module"></loadInformation124> <loadInformation227 model="Cisco IP Phone 7915 12-Button Line Expansion Module"></loadInformation227> <loadInformation228 model="Cisco IP Phone 7915 24-Button Line Expansion Module"></loadInformation228> <loadInformation229 model="Cisco IP Phone 7916 12-Button Line Expansion Module"></loadInformation229> <loadInformation230 model="Cisco IP Phone 7916 24-Button Line Expansion Module"></loadInformation230> <loadInformation30008 model="Cisco IP Phone 7902"></loadInformation30008> <loadInformation20000 model="Cisco IP Phone 7905"></loadInformation20000> <loadInformation369 model="Cisco IP Phone 7906"></loadInformation369> <loadInformation6 model="Cisco IP Phone 7910"></loadInformation6> <loadInformation307 model="Cisco IP Phone 7911"></loadInformation307> <loadInformation30007 model="Cisco IP Phone 7912"></loadInformation30007> <loadInformation30002 model="Cisco IP Phone 7920"></loadInformation30002> <loadInformation365 model="Cisco IP Phone 7921"></loadInformation365> <loadInformation484 model="Cisco IP Phone 7925"></loadInformation484> <loadInformation348 model="Cisco IP Phone 7931"></loadInformation348> <loadInformation9 model="Cisco IP Conference Station 7935"></loadInformation9> <loadInformation30019 model="Cisco IP Phone 7936"></loadInformation30019> <loadInformation431 model="Cisco IP Conference Station 7937"></loadInformation431> <loadInformation8 model="Cisco IP Phone 7940"></loadInformation8> <loadInformation115 model="Cisco IP Phone 7941"> SIP41.9-2-3S </loadInformation115> <loadInformation309 model="Cisco IP Phone 7941GE"> </loadInformation309> <loadInformation434 model="Cisco IP Phone 7942"> SIP42.9-2-3S </loadInformation434>
<loadInformation435 model="Cisco IP Phone 7945">SIP45.9-2-3S</loadInformation435> --identifies the
Avaya SIP Proxy and software to be used <loadInformation7 model="Cisco IP Phone 7960"></loadInformation7> <loadInformation30018 model="Cisco IP Phone 7961"></loadInformation30018> <loadInformation308 model="Cisco IP Phone 7961GE"></loadInformation308>
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
<loadInformation404 model="Cisco IP Phone 7962"></loadInformation404> <loadInformation436 model="Cisco IP Phone 7965"></loadInformation436> <loadInformation30006 model="Cisco IP Phone 7970"></loadInformation30006> <loadInformation119 model="Cisco IP Phone 7971"></loadInformation119> <loadInformation437 model="Cisco IP Phone 7975"></loadInformation437> <loadInformation302 model="Cisco IP Phone 7985"></loadInformation302> <loadInformation12 model="ATA phone emulation for analog phone"></loadInformation12> </Default>
The highlighted lines in the sample file above offer all endpoints registering the SIP Proxy
address and which firmware to download. This information can also be entered in the
SEP<MAC>.cnf.xml for each individual handset.
4.2.2. SEP<MAC>.cnf.xml
The SIP<MAC>.cnf.xml file is individual to each device. The <MAC> relates to the MAC
address of the handset. This file contains settings that are unique to each handset, so a file is
needed for each individual handset being registered with the Session Manager
Below is a sample of the file used to register a Cisco7945 (ext 24008) with Session Manager.
This file can also be created as a text file initially and then renamed with a .cnf.xml extension.
The same format applies to Cisco 7941 and Cisco 7942.
<registerWithProxy>true</registerWithProxy> --force Cisco endpoint to register with Session
Manager </sipProxies> <sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled> --permit endpoint to use Conference feature <localCfwdEnable>true</localCfwdEnable> --Active CfwdAll key on handset
<callForwardURI>service-uri-cfwdall</callForwardURI> --Updates with information as
entered by the user under CfwdAll key , however function is not working – unable to establish correct format for
URI <callPickupURI>service-uri-pickup</callPickupURI> <callPickupGroupURI>service-uri-gpickup</callPickupGroupURI> <callHoldRingback>2</callHoldRingback>
<remotePartyID>false</remotePartyID> --Set to false, else information in SIP Invites will not be
recognised by the Avaya.
</sipStack> <sipLines>
--Programming of Line Keys—
<line button="1"> --1st
Line key <featureID>9</featureID> --Cisco feature ID 9 – indicates a lines key – do not change
<featureLabel>Ext 24009</featureLabel> --Label to appear on the screen against the line <proxy>USECALLMANAGER</proxy> --indicates which SIP Proxy to use; refer to the line. Refers to the line
earlier in the file <processNodeName>192.168.81.119</processNodeName>
<port>5060</port> --SIP Proxy port 5060 <name>24009</name> --User name <displayName>User 24009</displayName> --Display Name – may be used in some SIP invites <autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled> --Auto Answer - see notes below regarding
configuration of activation/deactivationfor these and other services. </autoAnswer>
<callWaiting>0</callWaiting> --Call Waiting Activated/Deactivated – see notes below regarding
configuration activation/deactivation for these and other services <authName>24009</authName> --Authorisation Name for registering with SIP Proxy – refer to
System Manager � User Management for settings. <authPassword>123456</authPassword> --Authorisation passoword for registering with SIP Proxy –
refer to System Manager � User Management for settings. <sharedLine>false</sharedLine>
<messagesNumber>23500</messagesNumber> --Sets the destination under the Messages
button on endpoint <ringSettingActive>5</ringSettingActive> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>true</callerNumber>
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
<preferredCodec>g711ulaw</preferredCodec> --configure preferred codec <softKeyFile>softkeyDefault_kpml.xml</softKeyFile> --Name of the file that contains the
softkey settings for the endpoint <dialTemplate>dialplan.xml</dialTemplate> --Name of the dialplan file that contains the digit
patterns the endpoint can dial.
<kpml>1</kpml>
<phoneLabel></phoneLabel> --controls the display in the top left corner of screen <stutterMsgWaiting>2</stutterMsgWaiting> <disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig> <dscpForAudio>184</dscpForAudio> <dscpVideo>136</dscpVideo> </sipProfile> <commonProfile>
<phonePassword>cisco</phonePassword> --password for accessing the configuration
menu under the settings button <callLogBlfEnabled>2</callLogBlfEnabled> </commonProfile>
<loadInformation>SIP45.9-2-3S</loadInformation> --firmware load the endpoint shouldbe
running. If changed to something newer and the handset is rebooted, it will force an upgrade of the firmware. <vendorConfig> <videoCapability>1</videoCapability> </vendorConfig> <versionStamp>0032339366147827</versionStamp> <userLocale>
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
<transportLayerProtocol>2</transportLayerProtocol> --sets the handset to use TCP </device>
Notes
<autoAnswerEnabled>2</autoAnswerEnabled>. The numerical character 2 in the
line Auto Answer Enabled dictates the following for this service “2=off and locked so it can't be
changed through the settings menu”
The auto answer setting can be 0, 1, 2 or 3.
0=off and locked so it can't be changed through the settings menu.
1=on and locked so it can't be changed through the settings menu.
2=off and locked so it can't be changed through the settings menu.
3=on and locked so it can't be changed through the settings menu.
None of these settings will allow you to change it through the phone's settings menu.
<callWaiting>0</callWaiting> The call waiting setting can be 0, 1, 2 or 3.
0=off but can be changed through the settings menu.
1=on but can be changed through the settings menu.
2=off and locked so it can't be changed through the settings menu.
3=on and locked so it can't be changed through the settings menu.
<proxy>USECALLMANAGER</proxy> This setting against the earlier refers to using the IP address entered against the field
<processNodeName> which is found near the beginning of the file. Earlier versions of the SIP software may require this entry to be the IP address of the SIP Proxy, rather than the
expression USECALLMANAGER.
<loadInformation>SIP45.9-2-3S</loadInformation> This line dictates the software the handset should be using. The name is taken from the relevant
.loads file found in the firmware zip download, but without the extension label. This field can
also be used to upgrade a phones firmware at a later date.
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
It may be necessary to factory reset the Cisco Endpoint to clear it of all prior settings. To factory
reset 7941/42/45 do the following:-
• Remove the network cable.
• Press and hold down # whilst inserting the Network cable back in the correct port on the back of the handset. (The headset/mute/speaker keys may light up in turn)
• Keep holding the # key down until the line keys flash orange. • Key in 123456789*0#
• This will completely clear the handset, setting it back to factory defaults and the handset
will begin a search for a DHCP server. (Default Mode).
To perform a soft restart on the handset press Settings key in **#**. This will perform a reboot
of the handset without having to remove the network cable. This method does not factory reset
the handset. To perform a hard restart of the handset if the factory reset does not work at clearing
the handset configuration use the following procedure. Follow the instructions for factory reset
until the line keys are orange. Enter the code 3491672850*#. The handset screen will remain
dark throughout the entire process. Use Wireshark to monitor the handset for activity. After
approximately 10 minutes the handset will come back to normal boot procedures. Use with
caution and do not unplug the phone!
4.2.7. Configuring Fixed IP Address on Handset
If DHCP is not available the handset must be configured manually with IP information. The
process below details how to configure the IP information.
For SCCP Handset
• Press Settings
• Scroll down to 1-IPv4 Configuration
• Press **# to unlock config mode
• Make sure the line 1 DHCP Enabled is “highlighted”
• Press No – this should change 1 DHCP to Disabled
• Scroll to 2-IP Address press Edit and configure a fixed IP address
• Scroll to 3-Subnet Mask press Edit and configure a subnet mask
• Scroll to 4-Default Router 1 enter configure the default gateway address
• SAVE the changes
For SIP Handset
Repeat the above process but when entering **#, it will prompt for a password to unlock the
configuration. Enter the password cisco (all lowercase).
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
The Cisco endpoint should be programmed with a TFTP server address from which to download
the configuration and firmware files. The TFTP server address may either be issued via DHCP as
the phone registers for an IP address or the address may be added manually to the handset.
Configuration of TFTP via the handset (SCCP/SIP):
• Allow the handset to boot - the screen may show “Configuring IP” after a few minutes
• Press the Settings button
o If performing this activity on SIP phone, press **# to unlock the configuration. and use the password cisco to unlock the config prior to changing the TFTP
Server address.
• Press 2-Network Information
• Scroll down to 16-Alterate TFTP No
• For SCCP handset -Press **# to unlock the settings on the handset. (Padlock symbol will
show unlocked) and a YES key will appear on the screen.
• Press YES.
• The entry 16-Alterate TFTP should now show Yes.
• Press SAVE
• Scroll to 17-TFTP Server 1 and press Edit
• Enter the IP address of the TFTP server - use * to enter the decimals between the octets.
• Press VALIDATE to check the IP address
• Press SAVE to save the changes.
• The handset will attempt to contact the TFTP server to download any files.
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes
If the endpoint fails to come into service correctly, try the following:
• If the telephone symbol against the line key has an X against it, then settings are likely to
be incorrect and the handset has not fully registered. Check the entry in the
SEP<MAC>.cnf.xml for the <processNodeName>and check whether the <proxy1> entry is set to USECALLMANAGER. The <proxy1> may require the entry to be the IP
address of the SIP Proxy, rather than the expression USECALLMANAGER. Check the
fields in SEP<MAC>.cnf.xml for <authname> and line1_<authpassword> correspond to the names and passwords entered in the System Manager ����User
Management����Manage Users screens.
• Unable to dial other extensions – check the DIALPLAN.XML file to ensure the digit
patterns have been entered. Check the <dialTemplate> field in the SEP<MAC>.cnf.xml file is referencing the correct dialplan.xml file stored on the TFTP
server.
• Some information may be available in the handset via SETTINGS� 6-STATUS� 1-
STATUS MESSAGES.
5. Verification Steps This section provides details on how to verify Cisco handsets have registered successfully with
the Avaya Aura® Session Manager.
5.1. Verify Network Connectivity and Configuration File Download
Confirm via the Settings menu on the handset that a suitable IP address, default gateway and
subnet mask have been issued to the handset. Confirm via the Settings Menu that SIP details
have been issued to the handset
• Press Settings �
• 3-Device Configuration �
• Line 1 Settings
Review the fields underneath these options
o 1 Unified CM Configuraiton ���� 1 Unfied CM 1 – should show IP address of
SIP Proxy
o 2 SIP General Configuration ���� 1SIP General Configuration – shows codec
�1 Line Settings� Line 1 – shows the extn no.
All of these settings will be provided by the SEP<MAC>.cnf.xml. If an entry is missing or
incorrect, check the .xml files and reboot the handset to collect the updated files.
BD; Reviewed:
SPOC 03/25/2012
Solution & Interoperability Test Lab Application Notes