application notes for configuring dual avaya aura session ...€¦ · aura® communication manager...
TRANSCRIPT
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©2012 Avaya Inc. All Rights Reserved.
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Avaya Solution & Interoperability Test Lab
Application Notes for Configuring Dual Avaya Aura®
Session Manager R6.2 (Primary and Secondary) and Avaya
Aura® Communication Manager R6.0.1 with Cisco 7941G,
Cisco 7942G and Cisco 7945G Endpoints – Issue 1.0
Abstract
These Application Notes present a sample configuration for directly registering Cisco 7941G,
Cisco 7942G and Cisco 7945G as SIP devices with two Avaya Aura® Session Manager R6.2.
These two Avaya Aura® Session Managers function as a primary and secondary Active
Controllers. In the event the primary Avaya Aura® Session Manager goes out of service, the
secondary Avaya Aura® Session Manager will take over. The compliance testing focused on
Cisco handsets to register with more than one Avaya Aura® Session Manager and whether the
endpoints recognized the switchover to secondary Avaya Aura® Session Manager.
Additionally testing to establish call viability with Cisco endpoints whilst the switch between
Session Managers was being made as well as basic call feature support (Call Conference, Call
Hold).
Testing was conducted via the Internal Interoperability Program at the Avaya Solution and
Interoperability Test Lab.
NOTE: This Application Note is applicable with Avaya Aura® 6.2 which is currently in
Controlled Introduction. Avaya Aura® 6.2 will be Generally Available in Summer 2012.
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Table of Contents 1. Introduction ............................................................................................................................. 3
1.1. Testing Observations ........................................................................................................ 4
1.2. Equipment and Software Validated.................................................................................. 5
2. Configure Avaya Aura® Communication Manager ............................................................... 6
2.1. Verify Avaya Aura® Communication Manager License ................................................ 7
2.2. Administer System Parameter Features ........................................................................... 8
2.3. Administer IP Node Names.............................................................................................. 8
2.4. Administer IP Network Region and Codec Set ................................................................ 9
2.5. Administer SIP Signaling Group and Trunk Group ....................................................... 11
2.6. Administer Route Pattern ............................................................................................... 14
2.7. Administer Private Numbering ...................................................................................... 14
2.8. Administer Locations ..................................................................................................... 15
2.9. Administer Dial Plan and AAR Analysis ....................................................................... 15
2.10. Administer SIP Stations.............................................................................................. 16
2.11. Save Changes .............................................................................................................. 16
3. Configure Avaya Aura® Session Manager ........................................................................... 17
3.1. Log in to Avaya Aura® System Manager ...................................................................... 17
3.2. Administer SIP Domain ................................................................................................. 19
3.3. Administer Locations ..................................................................................................... 20
3.4. Administer Adaptations.................................................................................................. 22
3.5. Administer SIP Entities .................................................................................................. 22
3.6. Administer SIP Entity Link ............................................................................................ 26
3.7. Administer Time Ranges ................................................................................................ 28
3.8. Administer Routing Policy ............................................................................................. 29
3.9. Administer Dial Pattern .................................................................................................. 30
3.10. Administer Avaya Aura® Session Manager ............................................................... 32
3.11. Add Avaya Aura® Communication Manager as an Evolution Server....................... 35
3.12. Administer SIP Users for Cisco SIP devices .............................................................. 40
4. Configure Cisco Endpoints ................................................................................................... 44
4.1. Overview of Cisco Endpoint Configuration .................................................................. 44
4.2. Configuration Files for Cisco 7941/7942/7945 .............................................................. 47
4.3. Installing the Cisco Handset ........................................................................................... 59
5. Verification Steps.................................................................................................................. 60
5.1. Verify Network Connectivity and Configuration File Download .................................. 60
5.2. Verify Registration with Avaya Aura ® Session Manager ............................................ 61
6. Conclusion ............................................................................................................................ 62
7. Additional References ........................................................................................................... 62
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1. Introduction The purpose of this interoperability Application Note is to validate Cisco 7941G, Cisco 7942G
and Cisco 7945G endpoints which are directly registered as a SIP device with dual Avaya Aura®
Session Manager R6.2, operating as primary and secondary Active Controllers.
Cisco 7941G,7942G and 794545G handsets are normally configured via xml configuration files
(xml) delivered to the handset via TFTP server. The location of the TFTP server can be
administered via DHCP or via direct entry on the handset screen. Within the configuration files,
which are linked to the handset by the MAC address of the device, are settings to program the
extension number of the device, the SIP Proxy Server for the handset to register with and a
number of other settings i.e. voicemail.
The configuration below shows two Avaya Aura® System Manager R6.2 with Avaya Aura®
Communication Manager R6.0.1 which are both connected to an Avaya Aura® System Manager
R6.2 via a SIP trunk. Specifically SIP Entity Links exist between the two Avaya Aura® Session
Managers and the Avaya Aura® Communication Manager. Testing focused on Cisco endpoint
registration with both Session Managers, behavior in the event of the primary Session Manager
going out of service/returning to service, call survivability and handset functionality. Handset
capability when registered directly with Session Manager has been covered in a previous
Application Note. See Section 7 reference [6] for details of this Application Note.
Figure 1: Network Diagram of Dual Avaya Aura® Session Manager, Avaya Aura®
Communication Manager using Avaya and Cisco SIP Endpoints
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1.1. Testing Observations
Testing was carried out using a mixture of Avaya SIP endpoints, Cisco 7960/40 and Cisco
41/42/45 devices. 67% of test cases passed, the failures being related mostly to Cisco endpoints
inability to automatically recognize the primary Avaya Aura® Session Manager going out of
service. Once the Cisco endpoint has been manually restarted and recognizes the secondary
Avaya Aura® Session Manager as the Active Controller the handsets functioned as per previous
Application Notes.
NOTE: Cisco only supports the Cisco models 7960 and 7940 being connected directly to a third
party PBX. Cisco do not support the model ranges Cisco7941/42/45 connected with a third party
PBX. However Cisco has made end of life announcements for 7941 and 7960/40 ranges and the
7942/45 range are the recommended alternatives.
1.1.1. Switching Active Controllers
The main areas of failure related to the Cisco handsets frequent failure to recognize the switch in
Active Controllers (from primary to secondary Avaya Aura® Session Managers and vice versa),
where as the Avaya SIP handsets automatically recognized this and switched accordingly.
Although the models Cisco 7941/7942/7945 would sometimes automatically re-register with the
secondary Avaya Aura® Session Manager, by no means can this be guaranteed. The Cisco
models 7940/7960 always required a restart.
In the event the primary Avaya Aura® Session Manager returned to service, the Cisco endpoints
also rarely recognized this event, due to the phones configuration only recognizing one Active
Controller and not seeing any Standby Controller. In this event the handsets always needed a soft
restart to register back with the primary Avaya Aura® Session Manager.
The Cisco handsets can be restarted via a short key code entry on the handsets to force through
the re-registration with the secondary Avaya Aura®Session Manager. This usually takes under a
minute. Once they have re-registered and recognized the secondary Avaya Aura® Session
Manager as the Active “Communication Manager”, the handsets function as per the previous
Application Note regarding Cisco handsets registered directly with the Avaya Aura® Session
Manager.
It is recommended that the TFTP server which hosts the configuration files for the Cisco
endpoints be available at all times in the event of the Cisco endpoint being restarted. It was
observed that endpoints did not always come back into service with their previous settings and
instead went into an “UNPROVISIONED” state if the TFTP server was not available.
Cisco handsets can register with two Avaya Aura® Session Managers and indicate which is
Active and which is Standby, however once switched over to the Standby (which now becomes
Active), the original Active Avaya Aura® Session Manager is not recorded as either Active or
Standby.
The Avaya Aura® System Manager does not report via the User Registration Screen in System
Manager whether the devices are AST and does not report that the handsets are registered with
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the secondary Avaya Aura® Session Manager, although the registration status is available via
the traceSM tool and visible in the configuration screens on Cisco endpoints. (Avaya endpoints
usually show registration with both primary and secondary Avaya Aura® Session Managers on
this screen, although only one will be marked AC – Active Controller)
1.2. Equipment and Software Validated
The following equipment and software were used for the sample configuration provided.
Equipment Software
Avaya S8800 Server Avaya Aura® Communication Manager R6.0.1
VSP: 6.0.3.3.3
CM_Simplex template 6.1.0.0.2350
Avaya S8800 Server Avaya Aura® System Manager R6.2
Software Update Revision No: 6.2.11.1721
Avaya S8800 Server
Avaya Aura® Session Manager R6.2.0.0.62011
Note – Two servers were used at this specification for
Primary and Secondary Session Managers
Avaya G650 Media Gateway TN2312BP HW15 FW054
TN799DP HW16 FW040
TN2602AP HW08 FW061
TN2224CP HW08 FW015
TN2464CP HW02 FW024
TN793CP HW17 FW010
Avaya S8800 Server Avaya Aura® Messaging 6.1 (6.1.0.0.26115)
Note: See below
Avaya Handset 9611G SIP – S96x1_SALBR6_0_2_v470.tar
Avaya Flare A175 SIP – SIP_A175_1_0_3_00008.tar
Cisco 7941G CP-7941G Load File SIP41.9-2-3S
Cisco 7942G CP-7942G Load File SIP42.9-2-3S
Cisco 7945G CP-7945G Load File SIP45.9-2-3S
Cisco 7940G 7900 Series Firmware P03S-08-9-00
Cisco 7960G 7900 Series Firmware P03S-08-9-00
Windows Server 2003 Standard
Edition Service Pack 2
Functioning as DHCP and TFTP server for Cisco phone
programming purposes
Note: This is not a full test of voicemail with Cisco endpoints, and should not be construed as
meaning the full voicemail services are available to Cisco endpoints registered with Avaya
Aura® Session Manager. The test covered basic voicemail functionality – i.e. access and retrieve
messages after MWI appears on Cisco endpoint.
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2. Configure Avaya Aura® Communication Manager This section provides details on the configuration of Communication Manager. All
configurations in this section are administered using the System Access Terminal (SAT). This
section provides the procedures for configuring Communication Manager on the following areas:
• Verify Avaya Aura® Communication Manager License
• Administer System Parameters Features
• Administer IP Node Names
• Administer IP Network Region and Codec Set
• Administer Signaling Group and Trunk Groups
• Administer Route Pattern
• Administer Private Numbering
• Administer Locations
• Administer Dial Plan and AAR Analysis
• Administer Stations
• Save Changes
The following assumptions have been made as part of this document:
• It is assumed that Communication Manager, System Manager and Session Manager have
been installed, configured, licensed. Refer to Section 7, references [1]-[4] for
documentation regarding these procedures
• Throughout this section the administration of Communication Manager is performed
using a System Access Terminal (SAT). The commands are entered on the system with
the appropriate administrative permissions. Some administration screens have been
abbreviated for clarity
• The user has experience of administering the Avaya system via both SAT and Web Based
Management systems
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2.1. Verify Avaya Aura® Communication Manager License
Usethe 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
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2.2. Administer System Parameter Features
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, two entries are needed for the two
Session Managers - sm62vl81 and 192.168.81.119 are entered for the primary Session Manager.
(The identity sm62vl81 is the hostname of Session Manager server and sm62vl82 and
192.168.82.119 for the secondary Session Manager).
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 [PRIMARY Session Manager] sm62vl82 192.168.82.119 [SECONDARY Session Manager]
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2.4. Administer IP Network Region and Codec Set
Use the change ip-network-region n command, where n is the network region number to
configure the network region being used. In the sample network, ip-network-region 1 is used.
For the Authoritative Domain field, enter the SIP domain name configured for this enterprise
and a descriptive Name for this ip-network-region. Set the Intra-region IP-IP Direct Audio and
Inter-region IP-IP Direct Audio to yes to allow for direct media between endpoints. Set the
Codec Set to 1 to use ip-codec-set 1.
change ip-network-region 1 Page 1 of 20 IP NETWORK REGION Region: 1 Location: 1 Authoritative Domain: mmsil.local Name: To Session Manager MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 1 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? y UDP Port Max: 65535 DIFFSERV/TOS PARAMETERS Call Control PHB Value: 46 Audio PHB Value: 46 Video PHB Value: 26 802.1P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5
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Use the change ip-codec-set n command to configure IP Codec Set paramenters where n is the
IP Codec Set number. In these Application Notes, IP Codec Set 1 was used as the main default
codec set. The standard G.711 and G729A codecs were selected.
• Audio Codec Set for G.711MU, G.711A and G.729A
• Silence Suppression: Retain the default value n
• Frames Per Pkt: Enter 2
• Packet Size (ms): Enter 20
Retain the default values for the remaining fields, and submit these changes.
add ip-codec-set 1 Page 1 of 2 IP Codec Set Codec Set: 1 Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711A n 2 20 2: G.711MU n 2 20 3: G.729 n 2 20 4: G.729A n 2 20
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2.5. Administer SIP Signaling Group and Trunk Group
2.5.1. SIP Signaling Group
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 primary Session Manager, and an additional signalling group 151 and Trunk
Group 151 is used to reach the secondary 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 can be 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
display signaling-group 151 SIGNALING GROUP Group Number: 151 Group Type: sip IMS Enabled? n Transport Method: tcp Q-SIP? n SIP Enabled LSP? n IP Video? y Priority Video? y Enforce SIPS URI for SRTP? y Peer Detection Enabled? y Peer Server: SM Near-end Node Name: clan Far-end Node Name: sm62vl82 Near-end Listen Port: 5070 Far-end Listen Port: 5070 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? y H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6
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2.5.2. SIP Trunk Group
To add a corresponding trunk group use the command add trunk-group n, where n is the trunk
group number. Trunk Groups 150 and 151 for the primary and secondary Session Managers
should be created.
• 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
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Show ANSWERED BY on Display? y
Navigate to Page 4 and enter 101 for the Telephone Event Payload Type and P-Asserted-
Identity for Identity for Calling Party Display.
add 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
Screenshot of Trunk Group 151 for connectivity between Communication Manager and the
secondary Session Manager. The remaining settings for Pages 3 and 4 may be set the same as
those used for Trunk Group 150.
display trunk-group 151 Page 1 of 21 TRUNK GROUP Group Number: 151 Group Type: sip CDR Reports: y Group Name: SM62VLAN82 COR: 1 TN: 1 TAC: 151 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: 151 Number of Members: 12
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2.6. Administer Route Pattern
Configure a route pattern to correspond to the newly added SIP trunk groups. Use the change
route-pattern n command, where n is the route pattern number. Configure this route pattern to
route calls to trunk group 150, and then to trunk group 151as 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: 151 0 0 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 next 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 trunks. 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. In order to allow these number patterns to use either trunk group,
alter the trunk group number . For contiguous trunk groups, the pattern may be entered as TG1-
TG2. (See sample below). If the trunk group numbers are not contiguous a separate entry per
extension code will be needed. change private-numbering 0 Page 1 of 2 NUMBERING - PRIVATE FORMAT Ext Ext Trk Private Total Len Code Grp(s) Prefix Len 5 23 150-151 5 Total Administered: 9 5 24 150-151 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
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2.8. Administer Locations
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
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Use the change aar analysis 0 command to configure an entry for the SIP phone extensions
which begin with 24. Use unku for call type.
change aar analysis 0 Page 1 of 2 AAR DIGIT ANALYSIS TABLE Location: all Percent Full: 1 Dialed Total Route Call Node ANI String Min Max Pattern Type Num Reqd 20 4 4 150 unku n 230 5 5 150 unku n 231 5 5 150 unku n 232 5 5 150 unku n 233 5 5 150 aar n 235 5 5 150 unku n 23998 5 5 150 unku n 23999 5 5 150 unku n 24 5 5 150 unku n 25 4 4 150 unku n 3 7 7 999 aar n 35 5 5 150 unku n 37 5 5 150 unku n 38000 5 5 199 aar n 38001 5 5 199 aar n
2.10. Administer SIP Stations
To create Avaya and Cisco SIP endpoints, please see Section 3.12.
2.11. Save Changes
Use the save translation command to save all changes.
save translation SAVE TRANSLATION Command Completion Status Error Code Success 0
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3. Configure Avaya Aura® Session Manager
This section provides the procedure for configuring dual Session Manager. For further reference
documents, refer to Section 8 of this document. The procedures include the following areas:
• Log in to Avaya Aura® Session Manager
• Administer SIP Domain
• Administer Locations
• Administer Adaptations
• Administer SIP Entities
• Administer Entity Links
• Administer Time Ranges
• Administer Routing Policies
• Administer Dial Patterns
• Administer Avaya Aura® Session Manager
• Add Avaya Aura® Communications Manager as an Evolution Server
• Administer SIP Users
All configuration is carried out using System Manager R6.2.
3.1. Log in to Avaya Aura® System Manager
Configuration is accomplished by accessing the browser-based GUI of System Manager, using
the URL https://<ip-address>/SMGR where <ip-address> is the IP address of System
Manager.
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The Home screen is divided into three sections with hyperlinked categories below.
The programming of the relationship between Session Manager and Communication Manager
takes place under the section Elements����Routing. Once within this section there are a number
of screens to work through in order to set up the relationship between Session Manager and
Communication Manager. The Welcome screen shows the order in which these screens should
be programmed although not all screens are necessary in this example.
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3.2. Administer SIP Domain
SIP domains are created as part of the Session Manager basic configuration. There will be at
least one which the Session Manager is the authoritative SIP controller. In these sample notes it
is mmsil.local. The Session Manager can also deal with traffic from other domains, so multiple
SIP domain entries may be listed.
The location of where you are currently in the system is listed at the top of the screen.
Underneath will be listed the domain(s) available in the system.
To create a new SIP Domain, from the Home (first screen available upon successful logon)
select the following; Home ���� Elements ���� Routing ���� Domains ���� Domain Management
and click New.
• Name Add a descriptive name
• Type Set to SIP
• Notes, Add a brief description in the Notes field
Click Commit to save.
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3.3. Administer Locations
Session Manager uses the origination location to determine which dial patterns to look at when
routing the call if there are any dial patterns administered for specific locations. Locations are
also used to limit the number of calls coming out of or coming into physical locations. This is
useful for those locations where the network bandwidth is limited. For this sample configuration,
one Location has been created which will reference the Session Manager location. Navigate to
Home ���� Elements ���� Routing ���� Locations. To create a new Location, click New.
In the General section,
• Name Add a descriptive name
• Notes add a brief description
Leave the settings for Overall Managed Bandwidth and Pre-Call Bandwidth Parameters, as
default unless advised to do otherwise.
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Continue scrolling down the screen until Location Pattern is displayed. In the Location
Pattern section, under IP Address Pattern enter IP addresses used to logically identify the
location(s). Under Notes add a brief description. Click Commit to save.
In the example above, IP addresses have been entered with a (*) wildcard to indicate a range.
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3.4. Administer Adaptations
Adaptations are used to manipulate digits in the SIP URI strings of incoming and outgoing calls.
These are sometimes needed to manipulate SIP information from interconnecting third party
PBX. However, for this sample configuration, an Adaptation is not required, so no example is
shown.
3.5. Administer SIP Entities
Each SIP device (other than Avaya SIP Phones) that communicates with the Session Manager
requires a SIP Entity configuration. This section details the steps to create SIP Entity for the two
Session Managers and Communication Manager Server.
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To create a SIP Entity for the primary Session Manager, browse to Home ���� Elements ����
Routing ���� SIP Entities and click New. In the General section:
• Name Add a descriptive name.
• FQDN or IP Address Add the IP Address of the target entity (Session Manager1).
• Type, Select Session Manager.
• Notes Add a brief description.
• Location, Click on the drop down arrow and select the location
created in Section 3.3.
• Time Zone Select the appropriate Time Zone.
• SIP Link Monitoring Set to Use Session Manager Configuration
Click Commit to save. A message will appear advising that “Entity Links can be added to the
record once the Entity has been saved”. Section 3.6 advises how to create Entity Links.
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Repeat the process to create a SIP Entity for the secondary Session Manager.
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To create a SIP Entity for the Communication Manager, browse to Home ���� Elements ����
Routing ���� SIP Entities and click New. In the General section,
• Name Add a descriptive name.
• FQDN or IP Address Add the IP Address of the target entity (Session Manager).
• Type Select CM.
• Notes Add a brief description.
• Location Click on the drop down arrow and select the location
created in Section 3.3.
• Time Zone Select the appropriate Time Zone.
• SIP Link Monitoring Set to Use Session Manager Configuration
Click Commit to save. A message will appear advising that “Entity Links can be added to the
record once the Entity has been saved”. Section 3.6 advises how to create Entity Links.
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3.6. Administer SIP Entity Link
A SIP Trunk between a Session Manager and a telephony system is described by an Entity Link.
The next step is to create a SIP Entity Link, which included the transport parameters to be used
for communications between the Session Managers and Communication Manager and between
the two Session Managers.
Create a SIP Entity Link for Session Manager 1 to the Communication Manager. Browse to
Home ���� Elements ���� Routing ���� Entity Links. Click New.
• Name Enter a suitable identifier e.g. SM1 to CM
• SIP Entity 1 Drop-down and select the appropriate Session Manager.
• Protocol Drop down and select TCP.
• Port Enter 5060.
• SIP Entity 2 Drop-down select the SIP Entity added previously, i.e. ComManager.
• Port Enter 5060.
• Trusted Set the field as ticked.
• Notes Add a brief description.
Click Commit to save.
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Once the Entity Links have been created, return to the SIP Entities screen and check to see if the
Entity Links have been assigned to the SIP Entities.
Entity Links assigned to SIP Entity “SessMan1_vlan81”
If the Entity Links have not been added to the SIP Entity automatically, click Add and assign the
Entity Link manually. Create a further two entity links between Session Manager 2 and
Communication Manager and between Session Manager 1 and Session Manager 2.
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3.7. Administer Time Ranges
Create a Time Range for LCR routing which defines when policies will be active. To create a
Time Range browse to Home ���� Elements ���� Routing ���� Time Ranges. Click New. Under
Name enter a suitable identified. Select which Days are to be included in the Range. Set a
suitable Start Time and End Time. This will be used in configuring the Dial Plan. In Session
Manager, a default policy (24/7) is available that would allow routing to occur at anytime. This
was used in the example network.
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3.8. Administer Routing Policy
To complete the routing configuration, a Routing Policy is created. Routing policies direct how
calls will be routed to a system. A routing policy must be created for the Communications
Manager. This will be associated with the Dial Pattern which will be created in the next step.
To create a Routing Policy to route SIP traffic to Communication Manager, browse to Home ����
Elements ���� Routing ���� Routing Polices. Click New.
• Name Enter a suitable identifier
• Notes Enter suitable description.
• SIP Entity as Destination Click on Select. Choose the appropriate SIP Entity
(Communication Manager) that is to be the call destination
• Time of Day Click Add and select a suitable time range if more than one
is programmed. Click Select to add this time range and
return to the main screen
Click Commit to save.
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At this stage the records are missing the Dial Pattern for non SIP devices which will be created
next (Section 3.9).
3.9. Administer Dial Pattern
As one of its main functions, the Session Manager routes SIP traffic between connected devices.
Dial Patterns are created as part of the configuration to mange SIP traffic routing which will
direct calls based on the number dialed to the appropriate system. In the sample network, 5 digit
extensions beginning 24 are designated as Avaya SIP handsets. Internally SIP devices that are
registered with the same Session Manager do not need a Dial Plan creating. For devices that are
non SIP (i.e. H323 or Digital) or are SIP devices registered on another Session Manager or with
another PBX will need a Dial Plan entry. Below shows an example of a dial plan pattern for an
Avaya H323 phone which uses the dial pattern 231XX.
To create a Dial Pattern for calls browse to Home ���� Elements����Routing����Dial Patterns.
Click New.
• Pattern Enter a dial string pattern e.g. 231
• Min Enter the minimum number of characters in the extension
• Max Enter the maximum number of characters in the extension
• SIP Domains From the drop down select ALL
Scroll down the screen to the Originating Locations and Routing Policies area.
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In the Originating Locations and Routing Policies section click on Add. In the new window
from the Originating Location tick Apply selected Routing Policies to All Originating
Locations. (See Section 3.3 for how to create the Location). In the Routing Policies section,
tick the Routing Policy this should apply to. (See Section 3.8 on how to create Routing Policies)
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3.10. Administer Avaya Aura® Session Manager
To complete the configuration, adding the Session Managers to System Manager will provide the
linkage between the System Manager and the two Session Managers. On the System Manager
Home screen, under Elements select Session Manager���� Session Manager Administration.
On the right hand side, under Session Manager Instances, click on New.
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A new window will open.
Under General:
• SIP Entity name Select the names of the SIP entity added for Session Manager
• Description Descriptive Comment
• Management Access Point Host Name/IP
Enter the IP address of the Session Manager management
interface (Eth0)
• Direct Routing to Endpoints
Set to Enable
Under Security Module
• SIP Entity Address IP Address of the SIP Entity (see Section 3.5)
• Network Mask Enter the network mask corresponding to the IP address of the
Session Manager
• Default Gateway Enter the IP address of the default gateway for Session Manager.
Use default values for the remaining fields.
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Repeat for the secondary Session Manager.
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3.11. Add Avaya Aura® Communication Manager as an Evolution Server
In order for Communication Manager to provide configuration and Evolution Server support to
SIP Phones when they register to Session Manager, Communication Manager must be added as
an application.
3.11.1. Create Avaya Aura® Communication Manager Instance
On the System Manager Home screen Elements, select Inventory ���� Manage Elements. Click
New. Click on the General Tab and enter detail in the following fields.
• Name Enter a Descriptive Name
• Type Set to Communication Manager
• Description Free text entry
• Node Set to IP Address for CM SAT Access
All other fields on this tab may be left with default settings.
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Click on the Attributes Tab and enter detail in the following fields.
• Login Login used for SAT access
• Password Password used for SAT access
• Confirm Password Password used for SAT access
• Node Set to IP Address for CM SAT Access
All other fields may be left with default settings. Click Commit to save changes.
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3.11.2. Create an Communication Manager Evolution Server Application
For Communication Manager Evolution Server support, further configuration of the Session
Manager is required. Once complete the Session Manager will support Avaya SIP phone
registration. Users are created through the Session Manager User Management screens. Session
Manager then creates corresponding stations on the Communication Manager Evolution Server.
Configuration of the Evolution Server Application via Session Manager is a two stage sequence,
with the Application being created first, followed by the Application Sequence. To configure
browse to: Home ���� Elements ���� Session Manager ���� Application Configuration ����
Applications. Click New. Under Name enter a suitable identifier. Under SIP Entity drop-down
select the SIP Entity of the Feature Server. Under Description enter a suitable description. Click
Commit to save.
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To configure the Application Sequences Configuration. Browse to: Home ���� Elements ����
Session Manager ���� Application Configuration ���� Applications Sequences. Click New.
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.
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3.11.3. Synchronize Avaya Aura® Communication Manager Data
On the System Manager Home screen, select Elements����Inventory ����Synchronization ����
Communication System. Select the appropriate Element Name and the select Initialize data
for selected devices. Then click on Now.
Note: This Process can take some time. Progress can be monitored by clicking on Refresh and
the current Sync Status column will display the status so far. Once synchronization has finished,
the column will display Completed.
3.11.4. Data Replication
Additionally with two Session Manager, Data Replication after the addition of two Session
Managers should also be reviewed, prior to progressing further. To check the status of replication
of data between the two Session Managers and System Manager from the Home screen select
Replication���� Replication. The status of the Replica Group of Session Managers will be
displayed.
To view the replication status of the individual Session Managers, select the group and click
View Replica Nodes
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3.12. Administer SIP Users for Cisco SIP devices
SIP Users must be added via System Manager and the details will be updated on the CM. This
process is applicable to both Avaya and Cisco endpoints. The further configuration of Cisco
endpoints will be covered in the next chapter. The example in this document shows the
configuration of a Cisco 7945 with extension number 24008. These instructions will also apply
to Cisco 7941 and Cisco 7942. On the System Manager screen select Users, and then select User
Management���� Manage Users. Click New. On the Identity tab enter the following
information and use defaults for other fields
• Last Name Enter a desired last name
• First Name Enter a desired first name
• Login Name Enter the desired phone [email protected] where the domain was
defined in Section 3.2
• Password Password for the user to log into System Manager (SMGR)
• Localized Display Name
Can either be left blank or a preferred name typed in. If left blank the
Display Name will form based on the information entered in the Last
Name and First Name fields. This also applies to the field Endpoint
Display Name.
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Next click on the Communication Profile tab
• Communication Profile Password Password entered by user when logging into a phone
• Confirm Password Repeat of the above password
Expand Communication Address and click New
• Type Set to Avaya SIP
• Fully Qualified Address Enter the extension number and set the Domain
Click Add when this information has been entered.
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Scroll down the screen and expand the section Session Manager Profile.
• Primary Session Manager Select the Session Manager from the drop down
• Secondary Session Manager Select the secondary Session Manager.
• Origination Application Sequence Select the Communication Manager Sequence
programmed in Section 3.11.2
• Termination Application Sequence Select the Communication Manager Sequence
programmed in Section 3.11.2
• Home Location Select the location programmed in Section 3.3
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Scroll down the screen and expand the section CM Endpoint Profile. Enter the following fields
and use defaults for the remaining fields.
• System Select the CM Entity
• Extension Enter a desired extension number
• Template Select a telephone type template
• Port Select IP
• Voicemail Enter the access number for voicemail (optional)
• Delete Endpoint on Unassign of Endpoint from User or on Delete User
Tick this field to delete the record from CM when deleting the
user in this screen.
Click on Commit save changes.
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4. Configure Cisco Endpoints This section will cover the preparation and configuration of Cisco handsets in order to register
with two Session Manager. It will cover factory reset of the handsets, converting the handsets
from SCCP to SIP firmware and configuration of the files needed to provide information to the
Cisco handset relating to the Avaya registration details.
4.1. Overview of Cisco Endpoint Configuration
The Cisco endpoint 7941/42/45 receives configuration changes via a number of .xml files
containing the relevant data. Some files are optional and others are mandatory in order to make
the handset work. Files are delivered to the handset via TFTP server. Below is a brief description
of each file.
Filename Description
SEP<MAC>.cnf.xml [REQUIRED]
This file offers the handset information on the SIP Proxy
server to use, as well as time zone settings, voicemail
location etc. This file is applicable to all 7941/42/45
handsets registering to use Session Manager
Dialplan.xml [REQUIRED]
This file contains the dial patterns the Cisco devices uses
to dial out. Without this file present, the handset cannot
make outbound calls
softkeyDefaultkpml.xml [REQUIRED] File can be used to control function soft
keys i.e. Conference, transfer, hold etc.,
XMLDefault.cnf.xml [OPTIONAL] File can be used to delivery SIP Proxy
address and specify firmware load to all handset types
registered.
CTLSEP<MAC>.tlv [OPTIONAL]
Normally used by the Cisco Call Manager to provide
handsets with details of Server Certificates required for
connectivity with Cisco PBX.
ITLSEP<MAC>.tlv [OPTIONAL]
Normally used by the Cisco Call Manager to provide
handsets with details of Server Certificates required for
connectivity with Cisco PBX
Please see Section 4.2 for a full explanation of the files needed to configure the handsets.
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4.1.1. Firmware Files
Cisco devices may need converting from SCCP to SIP prior to installation on the Avaya.
Firmware can be downloaded from www.cisco.com although a login account may be required in
order to download the files.
When stepping through the links to final software download screen, ensure you select the ZIP
version of the file as the contents will need to be extracted to a TFTP server. The .cop and .sgn
versions are used by Cisco Call Manager PBX and do not need to be downloaded.
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Once the zip file has been downloaded and unzipped, the contents of the SIP Firmware ZIP file
are usually as follows. (Contents may vary slightly dependant on firmware downloaded).
Below gives a description of the file types and their purpose:
• cmterm7945_7965-sip.9-2-3.zip Original zip file containing the firmware files
• SIP45.9-2-3S.loads Firmware file. Contains the name of the file that will need
to be referenced by handset as it upgrades to SIP software.
• term45.default.loads Firmware file referenced by the SIP45.9-2-3S.loads for
updating Cisco 7945 endpoints.
• term65.default.loads Firmware file referenced by the SIP45.9-2-3S.loads for
updating Cisco 7965 endpoints.
• apps45.9-2-3TH1-9.sbn Firmware file
• cnu45.9-2-3TH1-9.sbn Firmware file
• cvm45sip.9-2-3TH1-9.sbn Firmware file
• dsp45.9-2-3TH1-9.sbn Firmware file
• jar45sip.9-2-3TH1-9.sbn Firmware file
For the models Cisco 7941 and Cisco 7942 the term files will include a reference to 41 or 42 i.e.
term41.default.loads. Please download the relevant software for the handset type. Transfer these
files to a TFTP server that the Cisco endpoint will be able to contact.
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4.2. Configuration Files for Cisco 7941/7942/7945
As previously stated the Cisco 7941/42/45 all use .xml files to provide configuration information
to the endpoint. These files are downloaded from the TFTP server to the endpoint. These files are
as follows:-
• XMLDefault.cnf.xml (Optional)
• SEP<MAC>.cnf.xml
• softkeyDefault_kpml.xml
• Dialplan.xml
4.2.1. XMLDefault.cnf.xml
The XMLDefault file is optional as the information contained within the file can also be entered
in the SEP<MAC>.cnf.xml file. The file is used to provide the SIP Proxy address and to specify
which version of firmware a device type should be using.
<Default> <callManagerGroup> <members> <member priority="0"> <callManager> <ports> <sipPort>5060</sipPort> </ports> <processNodeName>192.168.81.119</processNodeName> --identifies the primary Session Managery </callManager> </member> <member priority="1"> <callManager> <ports> <sipPort>5070</sipPort> </ports> <processNodeName>192.168.82.119</processNodeName> --indentifies the secondary Session Manager </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>
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<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> <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 with the two Session
Managers 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.
<device>
<deviceProtocol>SIP</deviceProtocol> --set the device to SIP
<devicePool> <dateTimeSetting> <dateTemplate>M/D/YA</dateTemplate> <timeZone>GMT Standard/Daylight Time</timeZone> <ntps> <ntp priority="0"> <name>0.0.0.0</name> <ntpMode>unicast</ntpMode> </ntp> </ntps> </dateTimeSetting>
<callManagerGroup> <members>
<member priority="0"> -- Entry details for the primary Session Manager
<callManager> <ports>
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<sipPort>5060</sipPort> --set the SIP Port to 5060
</ports>
<processNodeName>192.168.81.119</processNodeName> --set the primary Session Manager
</callManager> </member>
<member priority="1"> -- Entry details for the secondary Session Manager <callManager> <ports>
<sipPort>5070</sipPort> --set the SIP Port to 5060 </ports>
<processNodeName>192.168.82.119</processNodeName> --set the secondary Session Manager
</callManager> </member> </members> </callManagerGroup> </devicePool> <sipProfile> <sipProxies>
<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>
<semiAttendedTransfer>true</semiAttendedTransfer> --permits attended transfers
<anonymousCallBlock>2</anonymousCallBlock> <callerIdBlocking>0</callerIdBlocking> <dndControl>2</dndControl> <remoteCcEnable>true</remoteCcEnable> </sipCallFeatures> <sipStack> <sipInviteRetx>6</sipInviteRetx> <sipRetx>6</sipRetx> <timerInviteExpires>180</timerInviteExpires> <timerRegisterExpires>3600</timerRegisterExpires> <timerRegisterDelta>5</timerRegisterDelta> <timerKeepAliveExpires>120</timerKeepAliveExpires> <timerSubscribeExpires>120</timerSubscribeExpires> <timerSubscribeDelta>5</timerSubscribeDelta> <timerT1>500</timerT1> <timerT2>4000</timerT2> <maxRedirects>70</maxRedirects>
<remotePartyID>false</remotePartyID> --Set to false, else information in SIP Invites will not be
recognised by the Avaya.
</sipStack>
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<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> <redirectedNumber>true</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay> </line> <line button="2"> <featureID>9</featureID> <featureLabel>24009</featureLabel> <proxy>USECALLMANAGER</proxy> <port>5060</port> <name>24009</name> <displayName>24009</displayName> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>0</callWaiting> <authName>24009</authName> <authPassword>123456</authPassword> <sharedLine>false</sharedLine> <messagesNumber>23500</messagesNumber> <ringSettingActive>5</ringSettingActive> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>true</callerNumber> <redirectedNumber>true</redirectedNumber> <dialedNumber>true</dialedNumber>
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</forwardCallInfoDisplay> </line> </sipLines> <enableVad>true</enableVad>
<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> <name>English_United_States</name> <langCode>en</langCode> </userLocale> <networkLocale>United_States</networkLocale> <networkLocaleInfo> <name>United_States</name> </networkLocaleInfo> <authenticationURL></authenticationURL> <directoryURL></directoryURL> <servicesURL></servicesURL> <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices> <dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>1</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.
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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 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.
4.2.3. softkeyDefault_kpml.xml
The softkeyDefault_kpml.xml programs the softkeys on the Cisco handset. It provides the end
user with keys for hold, transfer, redial, DND, conference etc. No changes should need to be
made to this file, but a copy has been included in this document. Please see Appendix.
4.2.4. Dialplan.xml
Although the Avaya controls the digit and dial patterns for routing calls the Cisco endpoints also
use a dial plan file which controls what number patterns the handsets can dial. Without this file,
the handsets cannot make any outbound calls. The sample below shows various entries for
accessing certain services such as the operator, external lines using 9 for an outside line
<DIALTEMPLATE> <TEMPLATE MATCH="0" Timeout="1" User="Phone"/> <!-- Local operator--> <TEMPLATE MATCH="9,011*" Timeout="6" User="Phone"/> <!-- International calls--> <TEMPLATE MATCH="9,0" Timeout="8" User="Phone"/> <!-- PSTN Operator--> <TEMPLATE MATCH="9,11" Timeout="0" User="Phone" Route="Emergency" Rewrite="9911"/> <TEMPLATE MATCH="w!" Timeout="1" User="PHONE" Route="Emergency" Rewrite="9911"/> <TEMPLATE MATCH="9,.11" Timeout="0" User="Phone"/> <!-- Service numbers --> <TEMPLATE MATCH="9,1.........." Timeout="0" User="Phone"/> <!-- Long Distance --> <TEMPLATE MATCH="9,......." Timeout="0" User="Phone"/> <!-- Local numbers --> <TEMPLATE MATCH="23..." Timeout="0" User="Phone"/> <!—Avaya Digital H323--> <TEMPLATE MATCH="24..." Timeout="0" User="Phone"/> <!—Avaya SIP--> <TEMPLATE MATCH="81...." Timeout="0" User="Phone"/> <!—Siemens H4K--> <TEMPLATE MATCH="55..." Timeout="0" User="Phone"/> <!—Cisco UCM--> <TEMPLATE MATCH="3..." Timeout="0" User="Phone"/> <!-- Corporate Dial plan--> <TEMPLATE MATCH="*" Timeout="15"/> <!-- Anything else --> <TEMPLATE MATCH="123#45#6" Timeout="0" User="Phone"/> <!-- Match `#' --> <TEMPLATE MATCH="12\*345" Timeout="0" User="Phone"/> <!-- Match * Char --> <TEMPLATE MATCH="7,...." TIMEOUT="0" Tone="Bellcore-Hold" /> <!-- Play Hold --> <TEMPLATE MATCH="7,123,...." TIMEOUT="0" Tone="Bellcore-Hold" Tone="Cisco-Zip" /> <!--Play Hold after 7, Play Zip Tone after 123--> </DIALTEMPLATE>
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<TEMPLATE MATCH="23..." Timeout="0" User="Phone"/> <!—Avaya Digital H323--> <TEMPLATE MATCH="24..." Timeout="0" User="Phone"/> <!—Avaya SIP-->
The two lines in the above file permit the Cisco extension to dial any number starting 23 or 24
followed by 3 characters indicated by … The timeout field indicates to start dialling immediately
and the User is described as “Phone” which will be added to the SIP information being sent.
Once these files have been created, add them to the TFTP server.
4.2.5. SCCP or SIP Mode
To check if the Cisco endpoint is using either SCCP or SIP software, press the Settings button on
the handset. This is indicated by the following symbol:
Once inside the menu system go to:
• 6-STATUS
• 3-FIRMWARE VERSION.
If the App Load ID file name contains SCCP, then the endpoint is currently SCCP and will need
reconfiguring to SIP. If the file name contains SIP, then handset is already SIP configured.
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4.2.6. Factory Reset Cisco Endpoint
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).
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4.2.8. Configuring a TFTP server
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.
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Configuration of TFTP via DHCP server
In the example used below, the DHCP server used was installed on Windows 2003 Server
Standard Edition. The TFTP server should be added as Option 150 to the Scope options.
• Open the DHCP settings windows
• Highlight the server, right click and select Set Predefined Options (action not shown)
In the Predefined Options and Values window that appears, click on the Add button.
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Fill out the following fields
• Name Enter a name for the Option
• Data Type Select IP Address
• Code Set to 150 (for Option 150)
• Description Enter a description
Click OK to save the record.
Enter the IP Address of the TFTP server in the Value field and press OK.
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Return to the main DHCP screen and highlight Scope Options. Right click and select Configure
Options. In the Scope Options window, scroll down and select Option 150. Press OK.
The DHCP window should now show the scope option 150 on the right hand side.
Factory reset the handset and the TFTP server address will be issued to the Cisco endpoint.
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4.3. Installing the Cisco Handset
With all the relevant files placed on the TFTP server and the handset supplied with an IP address
and TFTP server address, connect the network cable to the back of the handset. Use Wireshark or
similar tool to either monitor the handset port or the TFTP server to review the files being
downloaded to the handset.
The handset will contact the TFTP server and begin downloading files, including the new
firmware. The handset may restart after downloading the firmware, and then restart again after
installing the new firmware. After this stage it should then load the Avaya settings into
configuration.
TFTP client showing files being downloaded to handset. The process can take around 5-10
minutes to complete.
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4.3.1. Faults after Configuration
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 Configuration ���� 1 Unified CM 1 – should show IP address of
primary Session Manager. Status should show as “Active”. 2 Unified CM2 should
show the IP address of the secondary Session Manager and status should be
“Standby”
o 2 SIP General Configuration ���� 1SIP General Configuration – shows codec
� 3 Register with Proxy = yes
� 9 Backup Proxy should show
secondary SM
� 10 Backup Proxy Port
�2 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.
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5.2. Verify Registration with Avaya Aura ® Session Manager
Log on to Session Manager using https://<ip-address>/SMGR. Once connected go to
Elements����Session Manager����System Status����User Registrations. Review the screen for the
Cisco handsets registered.
Note: Cisco endpoints DO NOT show as AST Devices or show on this screen as registered with
the Secondary Session Manager. However their registration can be observed via Session
Manager traceSM tool.
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6. Conclusion The interoperability of Cisco 7941/42/45 endpoints registered as SIP devices with two Avaya
Aura® Session Manager R6.2 and Avaya Aura® Communication Manager R6.0.1 is viable,
however some manual intervention is required in the event of a primary Session Manager failure
to keep the Cisco endpoints functioning. Please see Section 1.1 for observations and more detail.
7. Additional References Product Documentation for Avaya Products may be found at http://support.avaya.com
[1] Administering Avaya Aura® Communication Manager 03-300509 Release 6.0 Issue 6.0
[2] Administering Avaya Aura® Communication Manager Server Options 03-603479
Release 6.0.1, Issue 2.2
[3] Administering Avaya Aura® Session Manager 03-603324 Release 6.1 Issue 1.0
[4] Maintaining and Troubleshooting Avaya Aura® Session Manager 03-603325 Release
6.1 Issue 4.1
[5] Application Note for configuring Avaya Aura® Session Manager R6.2 and Avaya
Aura® Communication Manager R6.0.1 with Cisco 7960G and Cisco 7940G Endpoints
– Issue 0.
[6] Application Note for Configuring Avaya Aura® Session Manager R6.2 and Avaya
Aura® Communication Manager R6.0.1 with Cisco 7941G, 7942G and Cisco 7945G
Endpoints – Issue 1.0
Product Documentation for Cisco Products may be found at www.cisco.com. A login account
may be necessary to access some areas of the Cisco website for downloading software, etc..
[7] Cisco IOS Voice Command Reference:
http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/123tvr.html. Has
some use in interpreting the information held in the configuration files used by Cisco
endpoints.
[8] A number of documents relating to registering Cisco endpoints with alternative pbx’s
may be found at www.voip-info.org.
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Appendix
Sample of softkeyDefault_kpml.xml
<softKeyCfg> <versionStamp>0032339366147827</versionStamp> <typeSoftKey> <softKeyDef keyID="Redial"> <tag>1</tag> <eventID>1</eventID> <helpID>301</helpID> </softKeyDef> <softKeyDef keyID="NewCall"> <tag>2</tag> <eventID>2</eventID> <helpID>302</helpID> </softKeyDef> <softKeyDef keyID="Hold"> <tag>3</tag> <eventID>3</eventID> <helpID>303</helpID> </softKeyDef> <softKeyDef keyID="Transfer"> <tag>4</tag> <eventID>4</eventID> <helpID>304</helpID> </softKeyDef> <softKeyDef keyID="CFwdALL"> <tag>5</tag> <eventID>5</eventID> <helpID>305</helpID> </softKeyDef> <softKeyDef keyID="<<"> <tag>8</tag> <eventID>8</eventID> <helpID>308</helpID> </softKeyDef> <softKeyDef keyID="EndCall"> <tag>9</tag> <eventID>9</eventID> <helpID>309</helpID> </softKeyDef> <softKeyDef keyID="Resume"> <tag>10</tag> <eventID>10</eventID> <helpID>310</helpID> </softKeyDef> <softKeyDef keyID="Answer"> <tag>11</tag> <eventID>11</eventID> <helpID>311</helpID> </softKeyDef> <softKeyDef keyID="Confrn"> <tag>13</tag>
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<eventID>104</eventID> <helpID>313</helpID> </softKeyDef> <softKeyDef keyID="Park"> <tag>14</tag> <eventID>14</eventID> <helpID>314</helpID> </softKeyDef> <softKeyDef keyID="Join"> <tag>15</tag> <eventID>15</eventID> <helpID>315</helpID> </softKeyDef> <softKeyDef keyID="MeetMe"> <tag>16</tag> <eventID>16</eventID> <helpID>316</helpID> </softKeyDef> <softKeyDef keyID="PickUp"> <tag>17</tag> <eventID>17</eventID> <helpID>317</helpID> </softKeyDef> <softKeyDef keyID="GPickUp"> <tag>18</tag> <eventID>18</eventID> <helpID>318</helpID> </softKeyDef> <softKeyDef keyID="RmLstC"> <tag>57</tag> <eventID>19</eventID> <helpID>319</helpID> </softKeyDef> <softKeyDef keyID="Barge"> <tag>67</tag> <eventID>21</eventID> <helpID>321</helpID> </softKeyDef> <softKeyDef keyID="cBarge"> <tag>81</tag> <eventID>32</eventID> <helpID>332</helpID> </softKeyDef> <softKeyDef keyID="DirTrfr"> <tag>77</tag> <eventID>28</eventID> <helpID>328</helpID> </softKeyDef> <softKeyDef keyID="Select"> <tag>78</tag> <eventID>29</eventID> <helpID>329</helpID> </softKeyDef> <softKeyDef keyID="Dial"> <tag>0</tag>
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<eventID>100</eventID> <helpID>0</helpID> </softKeyDef> <softKeyDef keyID="DND"> <tag>63</tag> <eventID>69</eventID> <helpID>369</helpID> </softKeyDef> </typeSoftKey> <softKeySets> <softKeySet id="Ring Out"> <softKey keyID="EndCall"></softKey> </softKeySet> <softKeySet id="Connected"> <softKey keyID="Hold"></softKey> <softKey keyID="EndCall"></softKey> <softKey keyID="Transfer"></softKey> <softKey keyID="Park"></softKey> <softKey keyID="Confrn"></softKey> </softKeySet> <softKeySet id="On Hold"> <softKey keyID="Resume"></softKey> <softKey keyID="NewCall"></softKey> </softKeySet> <softKeySet id="On Hook"> <softKey keyID="Redial"></softKey> <softKey keyID="NewCall"></softKey> <softKey keyID="CFwdALL"></softKey> <softKey keyID="PickUp"></softKey> <softKey keyID="GPickUp"></softKey> <softKey keyID="DND"></softKey> </softKeySet> <softKeySet id="Off Hook"> <softKey keyID="Redial"></softKey> <softKey keyID="EndCall"></softKey> <softKey keyID="CFwdALL"></softKey> <softKey keyID="PickUp"></softKey> <softKey keyID="GPickUp"></softKey> </softKeySet> <softKeySet id="Remote In Use"> <softKey keyID="Barge"></softKey> <softKey keyID="NewCall"></softKey> <softKey keyID="cBarge"></softKey> </softKeySet> <softKeySet id="Ring In"> <softKey keyID="Answer"></softKey> <softKey keyID="DND"></softKey> </softKeySet> <softKeySet id="Off Hook With Feature"> <softKey keyID="Redial"></softKey> <softKey keyID="EndCall"></softKey> </softKeySet> <softKeySet id="Digits After First"> <softKey keyID="<<"></softKey> <softKey keyID="EndCall"></softKey>
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</softKeySet> <softKeySet id="Connected Transfer"> <softKey keyID="EndCall"></softKey> <softKey keyID="Transfer"></softKey> </softKeySet> <softKeySet id="Connected Conference"> <softKey keyID="EndCall"></softKey> <softKey keyID="Confrn"></softKey> </softKeySet> <softKeySet id="Local Conferenced"> <softKey keyID="Hold"></softKey> <softKey keyID="EndCall"></softKey> </softKeySet> </softKeySets> </softKeyCfg>
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©2012 Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks
are the property of their respective owners. The information provided in these Application
Notes is subject to change without notice. The configurations, technical data, and
recommendations provided in these Application Notes are believed to be accurate and
dependable, but are presented without express or implied warranty. Users are responsible for
their application of any products specified in these Application Notes.
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya Solution &
Interoperability Test Lab at [email protected]