application notes for configuring dual avaya aura session ...€¦ · aura® communication manager...

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Solution & Interoperability Test Lab Application Notes

©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 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.


• 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.



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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


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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.



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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.



<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"/>  <TEMPLATE MATCH="9,011*" Timeout="6" User="Phone"/>  <TEMPLATE MATCH="9,0" Timeout="8" User="Phone"/>  <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"/>  <TEMPLATE MATCH="9,1.........." Timeout="0" User="Phone"/>  <TEMPLATE MATCH="9,......." Timeout="0" User="Phone"/>  <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"/>  <TEMPLATE MATCH="*" Timeout="15"/>  <TEMPLATE MATCH="123#45#6" Timeout="0" User="Phone"/>  <TEMPLATE MATCH="12\*345" Timeout="0" User="Phone"/>  <TEMPLATE MATCH="7,...." TIMEOUT="0" Tone="Bellcore-Hold" />  <TEMPLATE MATCH="7,123,...." TIMEOUT="0" Tone="Bellcore-Hold" Tone="Cisco-Zip" />  </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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™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks

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Notes is subject to change without notice. The configurations, technical data, and

recommendations provided in these Application Notes are believed to be accurate and

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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

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