Avaya 96x1 Series Using Instructions

Using Avaya 96X1 SIP Agent Deskphones with Avaya Aura ® Call Center Elite

Release 8.1

Issue 1

June 2019

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Contents

Chapter 1: Introduction

............................................................................................................ 6

Purpose .................................................................................................................................. 6

Chapter 2: Implementation checklist

...................................................................................... 7

Chapter 3: Administering Communication Manager

........................................................... 10

Communication Manager templates ........................................................................................ 11

Administration tips ................................................................................................................. 11

Configuring IP node names and addresses ............................................................................. 14

Administering the IP Network Region screen ........................................................................... 14

Administering the Feature-Related System-Parameters screen ................................................. 15

Administering Call Center Elite features .................................................................................. 15

Administering the Signaling Group screen ............................................................................... 15

Administering the Trunk Group screen .................................................................................... 16

Administering call routing ....................................................................................................... 17

Administering the Numbering-Private Format screen ................................................................ 18

Administering the Station screen ............................................................................................ 18

Administering the Off-PBX-Telephone Station-Mapping screen ................................................. 19

Synchronizing Communication Manager and System Manager data .......................................... 19

Adding users and stations ...................................................................................................... 19

Chapter 4: Avaya 96X1 SIP agent deskphones

.................................................................... 21

Differences between 96X1 SIP and 96X1 H.323 deskphones .................................................... 21

96X1 SIP agent deskphone feature support ............................................................................. 22

Agent login, logout, and work mode changes ........................................................................... 23

Communication Manager invoked changes ............................................................................. 24

Personal Profile Manager ....................................................................................................... 25

Scalability of 96X1 SIP agent deskphones ............................................................................... 25

Chapter 5: Troubleshooting

................................................................................................... 27

96X1 SIP agent deskphone troubleshooting ............................................................................ 27

Communication Manager troubleshooting ................................................................................ 27

Communication Manager denial events ............................................................................. 27

Session Manager troubleshooting ........................................................................................... 29

Troubleshooting scenarios ..................................................................................................... 29

Chapter 6: Resources

............................................................................................................. 31

Documentation ...................................................................................................................... 31

Finding documents on the Avaya Support website ............................................................. 32

Training ................................................................................................................................ 33

Viewing Avaya Mentor videos ................................................................................................. 33

Support ................................................................................................................................ 34

Glossary

................................................................................................................................... 35

June 2019 Using 96X1 SIP agent deskphones with Call Center Elite

Comments on this document? [email protected]

5

Chapter 1: Introduction

Purpose

This document describes how to use and set up Avaya Aura ® Call Center Elite to work with 96X1

SIP agent deskphones.

This document is intended for people who want to learn how to use Avaya Aura ® Call Center Elite with 96X1 SIP agent deskphones, including implementation engineers and system administrators.



6

Chapter 2: Implementation checklist

Use the following checklist to ensure that you have set up all components for use with the 96X1 SIP agent deskphones.

# Task Description

1

2

3

4

5

6

7

Install System Manager

Install Session Manager

Create an enrollment password for trust management

Administer Session Manager

Install Communication Manager

Install the Communication Manager templates

Use the Communication Manager

System Management Interface

(SMI) to complete the configuration tasks

For information about installing System

Manager, see Deploying Avaya Aura ®

System Manager on System Platform .

For information about installing Session

Manager, see Deploying Avaya Aura ®

Session Manager .

For information about enrolling passwords, see Deploying Avaya Aura

®

Session

Manager .

For information about administering

Session Manager, see Deploying Avaya

Aura ® Session Manager .

For information about installing

Communication Manager, see Avaya Aura ®

Communication Manager Special

Application Features .

For Communication Manager at the core, download CM_Duplex.ovf file from Product

Licensing and Delivery System (PLDS).

For remote Communication Manager, download the CM_SurvRemote.ovf file.

For information about installing the

Communication Manager templates, see

Avaya Aura ® Communication Manager

Special Application Features .

For information about configuring

Communication Manager, see Avaya Aura ®

Communication Manager Special

Application Features .

Table continues…



7

Implementation checklist

#

8

9

10

11

12

13

14

15

16

17

Task Description

Configure Communication Manager as an Evolution Server

Administer Communication

Manager

Synchronize Communication

Manager data

Add users and stations

For information about configuring

Communication Manager as an Evolution

Server, see Administering Avaya Aura ®

Communication Manager .

For information about the SAT administration commands, see

Administering Avaya Aura

®

Communication

Manager Server Options .

For information about synchronizing the

Communication Manager data with the

System Manager database, see

Administering Avaya Aura ® Communication


For information about adding users, see



Install the 96X1 SIP agent deskphones

Gain access to CRAFT procedures For information about accessing local procedures, see Installing and Maintaining

Avaya 9601/9608/9608G/9611G/9621G/

9641G/9641GS IP Deskphones SIP .

Configure SIP settings For information about configuring the SIP settings, see Installing and Maintaining

Avaya 9601/9608/9608G/9611G/9621G/


Configure the time server settings

For information about installing SIP phones, see Installing and Maintaining

Avaya 9601/9608/9608G/9611G/9621G/


Set the Site-Specific Option Number

(SSON)

For information about configuring time server settings, see Installing and

Maintaining Avaya 9601/9608/9608G/

9611G/9621G/9641G/9641GS IP

Deskphones SIP .

For information about setting SSON, see

Installing and Maintaining Avaya

9601/9608/9608G/9611G/9621G/9641G/

9641GS IP Deskphones SIP .

Administer the 96X1 SIP agent deskphone options

For information about administering the options, see Administering Avaya

Deskphone SIP for 9601/9608/9611G/

9621G/9641G .




8

#

18

Task

Administer the 96X1 SIP agent deskphones for survivability

Description

For information about system failover and survivability, see Administering Avaya

Deskphone SIP for 9601/9608/9611G/

9621G/9641G .



9

Chapter 3: Administering Communication

Manager

Before you begin

Communication Manager must connect with Session Manager in a non-IP Multimedia Subsystem

(IMS) signaling group for Call Center functionality.

Ensure that the field option in the IMS Enabled field on the SIP Signaling Group screen is n to prevent Feature Invocation Failure when an agent logs in to the system.

Verify the field settings and values in the following fields on the System-Parameters Customer-

Options screen:

• ISDN-PRI

• IP Trunk

• Expert Agent Selection (EAS)

• Maximum Administered SIP Trunks

• Maximum Off-PBX Telephones-OPS

Ensure that the field option in the DID/Tie/ISDN/SIP Intercept Treatment field is attd and the

Trunk-to-Trunk Transfer field is restricted .

About this task

You can use SIP agents when you administer Communication Manager as an evolution server.

As an evolution server, Communication Manager uses the full-call model. In the full-call model, call processing is a single step procedure, that is, Communication Manager processes the origination and termination parts of the call without a break.

For information about IMS signaling flow and full-call model, see Implementing End-to-End SIP .

Procedure

1. Change the dialplan analysis.

2. Change the Feature Access Codes (FAC) for Automatic Alternate Routing (AAR) and

Automatic Route Selection (ARS).

3. Add a SIP domain in each network region, and administer an IP node name for the IP address of the Session Manager Security Module.

4. Add a SIP signaling group for each Session Manager Security Module. For message interchange between Communication Manager and the SIP server, select y in the Peer

Detection Enabled field on the Signaling Group screen.



10

Communication Manager templates

5. Add a trunk group of type SIP for call routing from Communication Manager to Session

Manager.

6. Administer the route pattern, and add the route pattern number as a proxy route.

7. Administer private numbering plan.

8. Administer the uniform dial plan.

9. Administer AAR and ARS.

10. Save translations.

Communication Manager templates

Communication Manager runs on System Platform as a virtualized version called a solution template, which includes all the features supported by Communication Manager.

Avaya offers the following Communication Manager templates:

• Communication Manager Main or Core

- Duplex Communication Manager Main or Survivable Core

- Simplex Communication Manager Main or Survivable Core

- Embedded Communication Manager Main

• Communication Manager Survivable Remote

- Simplex Communication Manager Survivable Remote

- Embedded Survivable Remote

Note:

You must install System Platform on Avaya servers before installing the solution template.

For the simplex and duplex templates, use the following servers:

• Common server R2: Dell ™ PowerEdge ™ R620 and HP ProLiant DL360p G8

• Common server R3: Dell ™ PowerEdge ™ R630 and HP ProLiant DL360 G9

For the embedded templates, use the Avaya S8300E server.

Administration tips

Note:

For more information about SIP configuration and related administration, see Implementing

End-to-End SIP .

• Use System Manager to administer Communication Manager features to ensure that

Communication Manager synchronizes translation files with the System Manager database.



11

Administering Communication Manager

• Ensure that the field option in the Expert Agent Selection (EAS) Enabled field on the

Feature-Related System Parameters screen is y .

• Administer the SIP Endpoint Managed Transfer field on the Feature-Related System

Parameters screen as n . You must select n to prevent unexpected events, such as inaccurate reporting or loss of User-to-User Information (UUI).

• Communication Manager provides the SIP Agent functionality only with the Evolution Server configuration and not with the Feature Server configuration. Therefore, you must administer the IMS Enabled field as n for all SIP signaling groups. If the field option is y ,

Communication Manager sends a Feature Invocation Failure error message when an agent attempts to log in to the system.

• Add a domain on each IP Network Region screen for all 96X1 SIP agent deskphones within a network region. Ensure that the authoritative domain field on the IP Network Region screen is equal to the SIP domain. If the domain entry is blank, SIP agent deskphones register with

Session Manager but do not display agent buttons, such as auto-in , manual-in , aux , or acw .

• SIP agent deskphones must have dedicated SIP trunk groups for:

- Call traffic with service providers and other Communication Manager servers.

- Call signaling with Session Manager.

Furthermore, Call Management System (CMS) or Avaya IQ must not measure Off-PBX

Station (OPS) SIP trunk groups that carry signaling data because signaling data is inconsistent with the format of call traffic data. Sharing of SIP trunk groups and measuring of signaling data can lead to loss of call traffic data and reporting errors.

To prevent data loss and reporting errors, Communication Manager does not send station signaling-related messages to CMS or Avaya IQ. Furthermore, if CMS or Avaya IQ measure

OPS SIP trunk groups, Communication Manager logs a SIP OPTIM TG Meas Error denial event 5073.

Use private settings to ensure that SIP trunks for Off-PBX Telephone Integration and Mobility

(OPTIM) OPS signaling are from dedicated SIP trunk groups. To prevent agent extension manipulation, administer the trunk group as private and the number format on the route pattern as unk-unk .

- Add SIP trunk numbers on the Private Numbering and the Public Unknown Numbering screens.

- Do not add extra digits to station extensions and agent extensions on the Private

Numbering or the Public Unknown Numbering screens. The extra digits can cause

Session Manager to send calls to Communication Manager for further processing without terminating calls to SIP agent deskphones. If private unknown numbering modifies agent extensions, agents can log out and then log in to the system. However, agents cannot use feature buttons to change work modes.

- Ensure that Session Manager has two entities for Communication Manager: One for inbound call traffic and the other for agent deskphone traffic. All SIP trunks must be on the

Processor Ethernet (PE) interface.

- Administer different listen port numbers for inbound trunks and SIP agent deskphone signaling trunks. Ensure that each function has dedicated trunk groups. For more information, see Administering the Signaling Group screen .

- The Session Manager Profile configuration for SIP agent deskphones must point to the

Communication Manager entity that you define for OPS trunk (station) signaling in the



12

Administration tips

Origination Application Sequence field and the Termination Application Sequence field. For more information, see Administering the Signaling Group screen . Without this administration, Session Manager cannot use relevant signaling trunks for call delivery to

SIP agent deskphones. This administration is important for a configuration where you use the same Communication Manager entity for all traffic, but now you want to separate the entity into dedicated inbound call traffic and OPS signaling data.

- Administer sequenced applications for users based on whether the user is on the originating or on the terminating side of a call. When you design the sequence for applications that act on deskphones that Communication Manager Evolution Server controls, you must ensure that Communication Manager is the last application that you define on the origination side of a call and the first application on the termination side of the call. This sequence is critical because the evolution server uses the full-call model for call processing, and all origination and termination feature processing occurs on the origination side of the call.

For more information, see Implementing End-to-End SIP .

- For dedicated SIP trunk groups for deskphone signaling with Session Manager, administer the Measured field on the Trunk Group screen as none . This administration prevents

CMS or Avaya IQ reporting issues, including loss of reporting.

• Use the phone number as the primary handle when you administer Session Manager. Do not use alphabets.

• Ensure that the phone type on the Station screen is one of the 96X1SIPCC types, and assign an agnt-login button on the Station screen for each SIP agent deskphone.

• The following are some guidelines for Secure Real-time Transport Protocol (SRTP):

Use the Transport Layer Security (TLS) protocol for signaling between Session Manager and

SIP agent deskphones and between Communication Manager and Session Manager.

Connection to Avaya Aura ® Presence Services requires TLS.

However, if you must use Transmission Control Protocol (TCP), ensure that you add the following parameters in the SIP settings file:

- Administer ENABLE_OOD_MSG_TLS_ONLY as 0. If you administer

ENABLE_OOD_MSG_TLS_ONLY as 1 , third-party call control Computer Telephony

Integration (CTI) applications and Supervisor Assist do not work.

- Administer the configuration parameter ENABLE_PPM_SOURCED_SIPPROXYSRVR as

0 . If the parameter is not 0 , the proxy server information from Personal Profile Manager

(PPM) overwrites the information in the settings file.

For more information, see Implementing End-to-End SIP .

• Configure the Simple Network Time Protocol (SNTP) server for SIP agent deskphones because Communication Manager checks the time on the certificate to validate the certificate.

• Administer media encryption parameters with the same cryptosuites on Communication

Manager and SIP agent deskphones. The default cryptosuite is aescm128–hmac80 .

• Administer none on the IP Codec Set screen for no encryption. The default setting is 9 that corresponds to the setting on the Communication Manager administration screen.

• Do not configure SIP agent deskphones with EC500 buttons because Communication

Manager as an evolution server does not support EC500.



13


Related links

Administering the Signaling Group screen on page 15

Configuring IP node names and addresses

About this task

If you do not configure IP node names, System Manager ignores the IP interfaces, which can result in unsuccessful IP synchronization of Communication Manager.

Perform the following procedure for each Survivable Core server, Survivable Remote server, and adjunct.

Procedure

1. At the command prompt, type change node-names ip . Press Enter .

2. In the Name field on the IP Node Names screen, type node names.

3. In the IP Address field, type the IP address.

4. Press Enter to save the changes.

Administering the IP Network Region screen

Procedure

1. At the command prompt, type change ip-network-region xxx , where xxx is the number of the network region. Press Enter .

2. In the Authoritative Domain field, select SIP .

3. In the Allow SIP URI Conversion field, select y .

4. In the Intra-region IP-IP Direct Audio field, select y .

5. Administer the Real-time Transport Control Protocol (RTCP) monitor server parameters.




14

Administering the Feature-Related System-Parameters screen

Administering the Feature-Related System-Parameters screen

Procedure

1. At the command prompt, type change system-parameters features and press

Enter .

2. In the Expert Agent Selection (EAS) Enabled field, select y .

3. In the SIP Endpoint Managed Transfer field on the Feature-Related System-Parameters screen, select n .

4. Administer the other fields that are relevant to the configuration in your organization.


Administering Call Center Elite features

The following are the primary screens for administering the Call Center Elite features:

• System-Parameters Customer-Options

• Feature-Related System Parameters

For information about administering all Call Center Elite features, see Administering Avaya Aura ®

Call Center Elite .

Administering the Signaling Group screen

Procedure

1. In the Group Type field, select SIP .

2. In the IMS Enabled field, select n .

3. In the Transport Method field, select tls .

4. Administer Near-end Listen Port and Far-end Listen Port . You must administer separate

SIP entities on Session Manager that map to different signaling groups in Communication

Manager to configure dedicated trunking for inbound trunks and for OPS station signaling trunks. Session Manager must have two entities for Communication Manager: One for inbound traffic and the other for agent phone traffic. For example, use the following settings for the Transport Layer Security (TLS) protocol.



15


Communication Manager side settings

Station signaling

Signaling-group 1: Set Near-end Listen Port to

5062 and Far-end Listen Port to 5062

Inbound trunk signaling

Signaling-group 2: Set Near-end Listen Port to

5061 and Far-end Listen Port to 5061

Session Manager side settings

SIP entities

CM1OPS, CMTRNK1, and SM1. Both the

CM1xx entities must point to the same

Communication Manager Fully Qualified

Domain Name (FQDN)

Entity links

• Entity Link 1: SM1 port 5062 to CM1OPS port

5062

• Entity Link 2: SM1 port 5061 to CM1TRNK port 5061

For TCP, use port 5060 instead of 5061.

Note:

With the sample configuration, signaling-group 1 and entity link 1 are logically pairedup to handle SIP OPS station signaling, and signaling-group 2 and entity link 2 are logically paired-up to handle inbound call signaling. However, you must administer appropriate routing in Session Manager, as well as in Communication Manager, to ensure that the appropriate traffic is routed over the appropriate signaling facilities. You can add additional signaling groups on Communication Manager by using the same entity links defined in Session Manager, as indicated in the table, as more OPS signaling trunks and more inbound signaling trunks might be required in

Communication Manager to handle the required call traffic load and the required number of SIP agent deskphones.

5. In the DTMF over IP field, select rtp-payload .

6. In the Session Establishment Timer field, enter 3 , which is the recommended time period for call center use.

7. In the Far-end Network Region field, assign a number from 1 to 250 that represents the region of Session Manager.

8. In the Far-end Domain field, assign the IP address of the SIP domain.

9. In the Initial IP-IP Direct Media field, select n . This field option is necessary to prevent unexpected interactions with the Call Center Elite features as some features depend on the media that Communication Manager handles during the first few seconds.

10. In the Direct IP-IP Audio Connections field, select y .

Administering the Trunk Group screen

The following procedure and settings refer to the trunk groups that are used only for SIP signaling

to the SIP agent deskphones. For more information, see Administration tips on page 11.



16

Administering call routing

Procedure

1. In the Group Type field, select SIP .

2. In the Signaling Group field, assign the number of the signaling group defined on the

Signaling Group screen.

3. In the Number of Members field, assign the number of 96X1 SIP phones that the trunk must support. Communication Manager automatically assigns the IP port numbers.

4. In the Redirect on OPTIM Failure (ROOF) field, enter 5000 (5 seconds).

5. In the Preferred Minimum Session Refresh Interval (sec) field, assign a value from 90 to 64800 . The default setting is 600 . To prevent spikes in the processor occupancy, assign a value that is greater than the sum of the average call handling time and the average call queuing time. For example, if the average call queuing time is 3 minutes and the average call handling time is 10 minutes, assign a field value that is greater than 780 seconds.

Note:

Although Communication Manager uses the configured value for outbound calls when the traffic is relatively low, during peak traffic conditions Communication Manager increases the session refresh interval value dynamically for protection against the processor occupancy spikes.

If the service provider network for SIP uses a session refresh interval value that is higher than the configured value for Preferred Minimum Session Refresh Interval but relatively low compared to the amount of SIP traffic that Communication Manager currently handles, Communication Manager rejects the calls with a 4xx response and includes the suggested session refresh interval value for call attempt. Therefore, large call centers must ensure that the service provider network uses a session refresh interval value that is greater than 1800 seconds within the INVITE message to ensure that Communication Manager accepts calls in the first attempt, regardless of the SIP call traffic load that Communication Manager might be handling.

6. In the Measured field, select none for the signaling trunks to Session Manager.

7. In the Numbering Format field, select private , which is the default for SIP.

8. In the Show ANSWERED BY on Display field, select y .

9. In the Support Request History field on the Protocol Variations page, select y . If the field setting is n , the SIP agent deskphone displays do not function correctly.

Administering call routing

Procedure

1. On the Locations screen, assign a proxy selection route pattern for locations that use

Session Manager.



17


2. On the IP Network Map screen, assign a calling party number for relaying to Public Safety

Answering Points (PSAPs) for 911 calls. Define the From and To IP addresses on the IP

Address Mapping screen to relay the correct emergency number.

3. On the Incoming Call Handling Treatment screen, define call handling for Session Manager trunk groups.

Administering the Numbering-Private Format screen

Private numbering plans ensure unique numbers for call routing. To administer the numbering plan, use the change private-numbering x command, where x is the extension length.

Administering the Station screen

Procedure

1. At the command prompt, type change station xxx , where xxx is the extension number. Press Enter .

2. In the Type field on the Station screen, select a 96X1SIPCC station type, that is,

9608SIPCC , 9611SIPCC , 9621SIPCC , or 9641SIPCC .

3. Assign a coverage path for each 96X1 SIP agent deskphone.

4. Administer the auto-answer field on the Station or Agent LoginID screen.

Note:

The field option on the Agent LoginID screen overrides the option on the Station screen.

5. In the Restrict Last Appearance field, select y .

6. Assign an agnt-login button along with agent work buttons and feature buttons, such as sip-sobsrv .

7. Administer the SIP feature options on page 6 of the Station screen. If you use an

Application Enablement Services (AES)-based application to control the agent deskphone, administer the Type of 3PCC Enabled field as Avaya . AES-based applications do not work if you do not administer this field.




18

Administering the Off-PBX-Telephone Station-Mapping screen

Administering the Off-PBX-Telephone Station-Mapping screen

About this task

The following procedure is optional. Administer the screen to change the default values.

Procedure

1. At the command prompt, type change off-pbx-telephone station-mapping xxx , where xxx is the extension number. Press Enter .

2. In the Application field, select OPS .

3. In the Mapping Mode field, select both .

4. In the Calls Allowed field, select all .

5. In the Bridged Calls field, select none .


Synchronizing Communication Manager and System

Manager data

Procedure

1. Click Elements > Inventory > Synchronization > Communication System .

2. On the Synchronize CM Data and Configure Options screen, expand the Synchronize CM

Data/Launch Element Cut Through table and the click the check box of the desired

Communication Manager Evolution Server.

3. Select Save Translations for selected devices and click Now to start the synchronization.

4. Refresh the Web page, and verify that the Sync Status column of the desired

Communication Manager Evolution Server shows Completed .

Adding users and stations

About this task

For each SIP user that you define in Session Manager, you must add a corresponding station in

Communication Manager. After administering a user, perform the following steps to automatically generate a corresponding SIP station.



19


Procedure

1. Click Elements > Inventory > Synchronization > Communication System .

2. On the Synchronize CM Data and Configure Options screen, click Launch Element Cut

Through .

If you log on to System Manager as an administrator, you do not require separate login credentials to gain access to the Element Cut Through screen. If you are a custom user, you must have the login credentials to gain access to the Element Cut Through screen.

3. On the Element Cut Through screen, enter the add station command.

Note:

If you administer stations directly on Communication Manager, you must administer a user communication profile for each extension.



20

Chapter 4: Avaya 96X1 SIP agent deskphones

9608, 9611G, 9621G, and 9641G are part of the multiline 9600 Series IP Deskphones. The deskphones are signaling protocol independent with two telephony applications, Avaya Deskphone

H.323 and Avaya Deskphone SIP, that support H.323 and SIP respectively.

9621G and 9641G are touch-based deskphones with a color display. 9611G and 9608 are buttonbased deskphones. 9611G has a color display while 9608 has a monochrome display. With the

9641G, 9608, and 9611G models, you can use a dual headset adapter so that two persons can listen to calls. You can also attach more than one Button Module (BM) to these deskphone models to extend call appearances, bridge appearances, or feature keys. For an agent, the deskphones offer convenient features and capabilities, including a phone screen to view and manage calls, and icons that indicate agent status, call state, feature status, queued calls, and missed calls.

Related links

Differences between 96X1 SIP and 96X1 H.323 deskphones

on page 21

96X1 SIP agent deskphone feature support on page 22

Agent login, logout, and work mode changes

on page 23

Communication Manager invoked changes

on page 24

Personal Profile Manager on page 25

Scalability of 96X1 SIP agent deskphones on page 25

Differences between 96X1 SIP and 96X1 H.323 deskphones

96X1 H.323

Communication Manager connection

H.323 deskphone connections are made on the line side of Communication Manager.

Server requirement

H.323 deskphones register with Communication

Manager.

Backup and restore

96X1 SIP

SIP deskphone connections are made on the trunk side of Communication Manager.

SIP deskphones register with Session Manager.




21

Avaya 96X1 SIP agent deskphones

96X1 H.323

96X1 SIP

H.323 deskphones use HTTP to store backup files.

SIP deskphones use Personal Profile Manager

(PPM) to store backup files.

Settings file and system parameters

The settings file is the same for the both types of deskphones, but some system parameters vary. In

H.323 deskphones, the OPSTAT and APPSTAT parameters control each user interface function.

In SIP deskphones, parameters such as

ENABLE_CONTACTS and ENABLE_CALL_LOG control each user interface function. In place of

APPSTAT, there are parameters such as

ENABLE_REDIAL, ENABLE_REDIAL_LIST,

ENABLE_MODIFY_CONTACTS,

ENABLE_CONTACTS and ENABLE_CALL_LOG.

In place of OPSTAT, there are parameters such as

PROVIDE_OPTIONS_SCREEN,

PROVIDE_LOGOUT, and

PROVIDE_NETWORKINFO_SCREEN.

Quality of Service (QoS)

H.323 deskphones use Communication Manager to set QoS.

Language support

SIP deskphones use parameters, such as L2QUAD,

L2QSIG, DSCPAUD, and DSCPSIG, to set QoS.

• H.323 deskphones do not support Hebrew and

Korean.

• SIP deskphones support text entry in Hebrew and

Korean.

• H.323 language files have a .txt file extension.

• All SIP language files have a .xml file extension.

Related links

Avaya 96X1 SIP agent deskphones on page 21

96X1 SIP agent deskphone feature support

The 96X1 SIP agent deskphones supports the following call center features:

• Agent login and logout buttons instead of Feature Access Codes (FACs): The phone shifts the button for login to logout after the agent logs in to the system.

• Auto and manual answer.

• Automatic display of collected digits with an incoming call that follows a call transfer.

• Auxiliary (AUX) work and After Call Work (ACW) buttons.

• Call Work Codes (CWCs) button.

• Correct messaging to the reporting adjuncts.

• Display of active Vector Directory Number (VDN).

• Display of Adjunct Switch Application Interface (ASAI) User-to-User (UUI) information.

• Entry of reason codes for change to the AUX work mode and for logout.



22

Agent login, logout, and work mode changes

• Hold, mute, transfer, conference, message waiting, elapsed call timer, date and time display, exit, and a minimum of three call appearances.

• Insertion of Vector Directory Number (VDN) of Origin Announcement (VOA) after answer with manual-answer operation, or accompanying zip tone for autoanswer operation.

• Interruptible AUX work.

• Manual-in and Auto-in work modes.

• Message Waiting Indicator (MWI) tracking for Expert Agent Selection (EAS) agent login IDs.

• Queue Status button: The q-calls button displays the number of calls in a queue and the time the oldest call is in a queue.

• Release button.

• Service Observing and Coaching: The sip-sobsrv button is used by supervisors to monitor and coach EAS agents when agents speak with callers. For coaching, supervisors need to press the Coach soft key on SIP CC endpoints. Observers can hear VOA only after they have joined the observed call.

• Stroke/Event Count button.

• Third-party MWI button.

• Visible and audible confirmation of feature activation and status change to the agent.

• VuStats button.

Note:

Agent Greetings is unavailable with the 96X1 SIP agent deskphones. The feature is supported only with phones that use the Avaya Deskphone H.323 application. Call Center features, such as login and logout, function differently in the 96X1 SIP agent deskphones as these features use the capabilities of the Avaya SIP architecture.

Related links


Agent login, logout, and work mode changes

The 96X1 SIP agent deskphones support the following basic call center features.

For more information about features and operations, see Using Avaya Deskphone SIP for

9608/9611G for Call Center Agents and Using Avaya Deskphone SIP for 9621G /9641G for Call

Center Agents .

Agent login and logout

A single login/logout button is available for agent login and logout. Once an agent logs in, the button toggles to logout. The logged in agent can view the skills associated with the login ID. If the agent is on an ACD call and presses logout , the phone lamp lights to indicate a pending logout.

You can also administer a requested or forced logout reason code to request or compel an agent to enter a reason code.



23


Agent work mode change

The auto-in and manual-in feature buttons are available for agents to change the work mode.

You can administer auto-in so that Communication Manager delivers calls to agents automatically.

In the manual-in work mode, the agent must receive calls manually.

Agent state change to Auxiliary (AUX) Work

The aux-work feature button, if administered as forced or requested , is available with entry of an AUX Work reason code.

If an agent is on an ACD call and presses aux-work , Communication Manager accepts the request for change of agent work state and displays a pending indication on the phone display until the agent drops the call. Communication Manager then notifies the agent of the work state change.

Agent state change to After Call Work (ACW)

When an agent is on an ACD call and presses acw , Communication Manager accepts the request for change of agent work state and displays a pending indication on the phone display until the agent drops the call. Communication Manager then notifies the agent of the work state change.

Related links


Communication Manager invoked changes

Communication Manager notifies the deskphone of changes to the agent work state, agent login, or logout to account for situations such as the following:

• When an agent state automatically changes to ACW after releasing or disconnecting the call,

Communication Manager notifies the SIP phone of an agent state change to the Manual-in work mode.

• When an agent in the Auto-in work mode disconnects an ACD call, Communication Manager notifies the SIP phone of an agent state change to Timed ACW.

• When Maximum Agent Occupancy (MAO) is less than the threshold, Communication

Manager notifies the SIP phone of an agent state change from AUX work to Available. The

SIP phone displays the reason code for the state change.

• When the administered forced logout from ACW or clock time for an agent is reached,

Communication Manager plays a tone if the agent is on an ACD call. The agent can press logout-ovr to cancel the forced logout. If the agent does not press logout-ovr , a pending logout indication displays on the SIP phone and Communication Manager logs the agent out after the agent disconnects the ACD call.

• When you administer an agent AUX work mode as interruptible, Communication Manager notifies the SIP phone if the agent state changes from AUX work to Available.

• When agents in a particular skill or location are forced to logout or enter the AUX Work mode,

Communication Manager notifies the SIP phone.

Related links




24

Personal Profile Manager

Personal Profile Manager

Personal Profile Manager (PPM) downloads SIP phone-related data from the local Session

Manager server to the 96X1 SIP agent deskphones. When data changes in Communication

Manager, Session Manager sends a PUBLISH or NOTIFY message to the 96X1 SIP agent deskphones. The deskphones then request PPM for the updated data.

PPM downloads the following call center feature buttons in addition to the other basic functions:

• after-call with data for Grp

• agnt-login

• auto-in with data for Grp

• auto-msg-wt with the extension number

• aux-work with Auxiliary (AUX) data for RC and Grp

• logout-ovr

• manual-in

• q-calls with data for Grp

• sip-sobsrv with data for Listen-Only?

and Coach options

• stroke-cnt code

• uui-info

• vu-display with aux data for Fmt

• work-code

After an agent logs in to the system, the 96X1 SIP agent deskphones send a SUBSCRIBE message to PPM or SM to download agent characteristics through the Call Center Information

(CC-Info) Event package. The package includes agent information (AgentInfo) and CC statistical information (CCStatsInfo).

Note:

PPM does not override the values that you set using the CRAFT menu on the phone. You must manually clear the values.

Related links


Scalability of 96X1 SIP agent deskphones

• The Dell ™ PowerEdge agent deskphones.

™ R620 and HP ProLiant DL360p G8 servers support up to 10,000 concurrently logged-in SIP Expert Agent Selection (EAS) agents who use the 96X1 SIP

• The Dell ™ PowerEdge agent deskphones.

™ R630 and HP ProLiant DL360 G9 servers support up to 10,000 concurrently logged-in SIP Expert Agent Selection (EAS) agents who use the 96X1 SIP



25


Related links




26

Chapter 5: Troubleshooting

96X1 SIP agent deskphone troubleshooting

For information about troubleshooting the 96X1 SIP agent deskphones, see Installing and

Maintaining Avaya 9601/9608/9608G/9611G/9621G/9641G/9641GS IP Deskphones SIP .

Communication Manager troubleshooting

Use the following list trace commands to capture information on a specific station or trunk:

• list trace sip-station <extension number>

• list trace station <station number>

• list trace tac <tac number>

You can use the list trace commands to troubleshoot the following:

• Misdirected calls

• Call denials

• Trunking and routing problems

• DS1 connectivity to other vendor equipment

Communication Manager denial events

Event type Event description

374 Suspend vectors in ovld

375

1039

Resume vectors after ovld

ACD login failed

Explanation

When the processor occupancy of Communication Manager exceeds 92.5%, the system suspends the vectors for 6 seconds.

The system re-checks the processor occupancy and if the processor occupancy has not still reduced, the system continues to suspend the vectors processing after every 6 seconds.

When the processor occupancy of Communication Manager reduces, the system resumes the vectors processing.

Group Manager or User Manager set up of the ACD Logical Agent login information failed before password matching, if any.




27

Troubleshooting


1363 SIP Agent logins maximum

1375

1380

1381

1382

1383

1384

1385

1386

1387

1388

2120

2127

5073

5077

5078

Double agent login to station

Agent login failure

Agent login failure

Explanation

Maximum number of simultaneous SIP EAS Agents logins exceeded.

Agent is logging in to a physical station that has another agent already logged in.

Agent login failure in getting the number of digits in the Logical

Agent password. The system cannot find the login ID or user ID or the ID is invalid.

Possible causes:

• An agent who logs in to a Multiple Call Handling (MCH) split or adjunct-controlled split is already logged in to the system.

• The Expert Agent Selection (EAS) field on the Feature-Related

System-Parameters screen is n .

Login is invalid.

Agent login invalid/ error

Agent login failure/ error

Logical Agent failure in getting the agent login ID. Possible causes are as follows:

• Error in initializing agent-stat table.

• Login for the skill failed.

• Logging in to skill that the agent has already logged in to before.

Agent logins maximum

Agent password digits failed

Agent password mismatch

Agent login invalid/ error

• Maximum number of logged in skill reached.

Maximum number of simultaneous logins exceeded or agent login failed.

Failure in getting the Logical Agent password digits from the Dial

Plan Manager.

Agent entered a password that does not match the administered password.

Login is invalid.

Login acceptance fails Logical Agent login processing of agent login messages failed.

Advocate agents exceed maximum

Maximum number of Business Advocate agents already logged in.

Reached maximum BCMS capacity.

Over BCMS agent login cap

SIP OPTIM TG Meas

Error

SO-Coach-In socoach mode

SO-Coach-not reached agnt

Trunk groups for SIP OPTIM OPS signaling are defined as measured and SPI events have been blocked.

Cannot toggle between Service Observing Listen Only and Listen

Talk modes while Coaching is activated.

Cannot activate Coaching until call connects to an agent.




28

Session Manager troubleshooting


5079 SO-Coach-already active

5080

5081

SO-Coach-invalid notalk

Unsupported CMS release

5082

5083

5084

5085

Unsupported AAPC release

SO-Coach-In conference

SO-Coach-in waitstate

SO-Coach-not servobsrvng

Explanation

The maximum number of coaches are on the call: one.

Cannot coach in the Service Observe No Talk mode.

CMS release read from PREC is no longer supported and was blanked out.

AAPC release read from PREC is no longer supported and was blanked out.

Cannot coach during conferences.

Cannot coach until Service Observer is active on a call.

Service Observing must be activated in order to use Coaching.

Session Manager troubleshooting

Use SIP message tracing to troubleshoot Session Manager instances. The SIP Trace Viewer displays SIP message trace logs based on the configured filters.

For information about SIP tracing, see Maintaining and Troubleshooting Avaya Aura ®

Manager .

Session

Troubleshooting scenarios

Problem

You cannot register stations.

Troubleshooting actions

• Check that the Communication Manager signaling group and

Session Manager media server have a consistent media type

(TCP/TLS).

• Check that the Off-PBX-Telephone Station-Mapping screen has the correct trunk.

• Check that the deskphone has the correct sip-server IP address.




29

Troubleshooting

Problem

The Phone is registered, but the feature buttons are not available or are not working.

An agent cannot log in.

The 96X1 SIP agent deskphone does not support third-party call forwarding and send all calls.

Multiple call appearances on an incoming call.

Troubleshooting actions

• Ensure that you have added a domain to all the IP network regions associated with the 96X1 SIP agent deskphones. For more

information, see Administration tips on page 11.

• Restart PPM via service tomcat4 restart. If restarting does not resolve the issue, set Session Manager using stop -acfn; start ac /var/log/sip-server/ppm.log.

• Look at the denial events using display events with type = denial.

• Check that the deskphone is administered as a 96X1SIPCC station type on the Station screen.

• On the Call Center pages of the System-Parameters Customer screen, verify that the Logged-In SIP EAS Agents field is greater than 0.

• Check that the subscriptions are set up between Communication

Manager and Session Manager. Use tcm and enter rdd :sus

Vmem . Navigate to the next page. Verify that the cAgentStatusSub number equals the number of signaling groups going to a Session

Manager.

• Other errors might be existing agent errors, such as multiple agent login and incorrect login ID and password.

Check that administration of third-party support for call forwarding and send all calls exists. When buttons are administered for call forwarding all , call forwarding busy/does not answer , or send all calls , leave the corresponding extension fields on the feature button assignments portion of the Station screen blank.

On page 2 of the Off-PBX-Telephone Station-Mapping screen, verify that the Bridged Calls field is set to none . The field must be set to none for any SIP station that has bridged to the station. For instance, consider three SIP stations in this scenario:

• SIP station A is administered with three primary call appearances and one bridged appearance for SIP station B.

• SIP station C is administered with three primary call appearances and two bridged appearances for SIP station A. Administer the

Bridged Calls field for all phones to none .



30

Chapter 6: Resources

Documentation

See the following related documents.

Table 1: Avaya Aura ® Session Manager documents

Title

Deploying Avaya Aura

®

Session Manager

Use this document to:

Deploy Session Manager.

Audience

Implementation engineers and system administrators

All users Maintaining Avaya Aura

®

Session Manager

Troubleshooting Avaya Aura ® Session

Manager

Table 2: Call Center Elite documents

Maintenance tasks for the system.

Resolve basic problems of the system.

All users

Title Use this document to:

Administering Avaya Aura ® Call Center Elite Administer Call Center Elite features.

Avaya Aura ® Call Center Elite Overview and

Specification

Know about Call Center Elite features, performance specifications, security, and licensing information.

Avaya Aura ® Call Center Elite Feature

Reference

Know about Automatic Call

Distribution (ACD) and Call

Vectoring features.

Audience


Implementation engineers, sales engineers, and solution engineers

All users

Table 3: Communication Manager documents

Title


Manager

Avaya Aura ® Communication Manager Denial

Events


Administer Communication

Manager.

Resolve basic problems and maintain the system.

Audience


All users



31

Resources

Table 4: SIP Agent Deskphones documents

Title

96X1 SIP Setting File Parameters SIP 6.2 or later

Avaya Deskphone SIP 9600 Series IP deskphones - API Guide

Avaya 9608/9608G/9611G IP Deskphones

SIP Quick Reference


Administer 96X1 SIP agent deskphones.

Develop APIs.

Avaya Deskphone SIP for 9621G/9641G

Quick Reference

Installing and Maintaining Avaya

9601/9608/9608G/9611G/9621G/9641G/

9641GS IP Deskphones SIP

Using Avaya 9608/9608G/9611G IP

Deskphones SIP for Call Center Agents

Using Avaya 9621G/9641G/9641GS IP

Deskphones SIP for Call Center Agents

Using Avaya 9608/9608G/9611G IP

Deskphones SIP

Using Avaya 9621G/9641G/9641GS IP

Deskphones SIP

Audience


API developers

Read about the 9608 and

9611G SIP agent deskphones.

Read about the 9621G and

9641G SIP agent deskphones.

Install, configure, administer, and maintain 96X1 SIP agent deskphones.

Use the 9608 and 9611G SIP agent deskphones in a call center environment.

Use the 9821G and 9641G

SIP agent deskphones in a call center environment.

Use the 9608 and 9611G SIP agent deskphones.

Use the 9621G and 9641G

SIP agent deskphones.

All users

All users


All users

All users

All users

All users

Table 5: Other related documents

Title

Deploying Avaya Aura ® System Manager on

System Platform

Implementing End-to-End SIP


Deploy Avaya Aura ® System

Manager.

Configure SIP endpoints.

Audience



Finding documents on the Avaya Support website

Procedure

1. Go to https://support.avaya.com

.

2. At the top of the screen, type your username and password and click Login .

3. Click Support by Product > Documents .

4. In Enter your Product Here , type the product name and then select the product from the list.



32

Training

5. In Choose Release , select an appropriate release number.

6. In the Content Type filter, click a document type, or click Select All to see a list of all available documents.

For example, for user guides, click User Guides in the Content Type filter. The list displays the documents only from the selected category.

7. Click Enter .

Training

The following courses are available on www.avaya-learning.com

. Enter the course code in the

Search field, and click Go to search for the course.

Course code

ACIS-7391

73600V

7391X

Course title

Implementing Avaya Aura ® Call Center Elite 40 hours

Avaya Aura ® Call Center Elite and Avaya Aura ® Call Center Elite Multichannel

Implementation Exam 1.50 hours

ACSS-7491

74600V

7491X

2416W

2412W

2414W

Supporting Avaya Aura ® Call Center Elite 16 hours

Avaya Aura ® Call Center Elite and Avaya Aura ® Call Center Elite Multichannel

Support Exam 1.50 hours

Avaya Aura ® Call Center Elite Fundamentals 0.5 hour for all audiences

Using Avaya Workspaces for Elite – Agents 0.5 hour for end-users

Using Avaya Workspaces for Elite – Supervisors 0.5 hour for end-users

Viewing Avaya Mentor videos

Avaya Mentor videos provide technical content on how to install, configure, and troubleshoot

Avaya products.

About this task

Videos are available on the Avaya Support website, listed under the video document type, and on the Avaya-run channel on YouTube.

Procedure

• To find videos on the Avaya Support website, go to https://support.avaya.com/ and do one of the following:

- In Search , type Avaya Mentor Videos to see a list of the available videos.



33

Resources

- In Search , type the product name. On the Search Results page, select Video in the

Content Type column on the left.

• To find the Avaya Mentor videos on YouTube, go to www.youtube.com/AvayaMentor and do one of the following:

- Enter a key word or key words in the Search Channel to search for a specific product or topic.

- Scroll down Playlists, and click the name of a topic to see the available list of videos posted on the website.

Note:

Videos are not available for all products.

Support

Go to the Avaya Support website at https://support.avaya.com

for the most up-to-date documentation, product notices, and knowledge articles. You can also search for release notes, downloads, and resolutions to issues. Use the online service request system to create a service request. Chat with live agents to get answers to questions, or request an agent to connect you to a support team if an issue requires additional expertise.



34

Glossary

AAR adjunct

AES appearance application

ARS

ASAI auto-in

AUX work

When resources are unavailable, Communication Manager uses the

Automatic Alternate Routing (AAR) feature to route calls to a different route than the first-choice route.

A processor that does tasks for another processor and is optional in the configuration of the other processor. See also

application

on page 35.

Application Enablement Services (AES) is an Avaya product that provides a platform for the development of CTI-based applications for

Communication Manager.

A software process that is associated with an extension and whose purpose is to supervise a call. An extension can have multiple appearances. Also called call appearance, line appearance, and occurrence.

An adjunct that requests and receives ASAI services or capabilities.

Applications can reside on an adjunct. However, Communication

Manager cannot distinguish among several applications residing on the same adjunct. Hence, Communication Manager treats the adjunct and all resident applications as a single application. The terms application and adjunct are used interchangeably throughout the document.

Automatic Route Selection (ARS) is a feature that Communication

Manager uses to automatically select the least cost route to send a toll call.

Adjunct-Switch Application Interface (ASAI) is an Avaya protocol that applications use to gain access to the call-processing capabilities of

Communication Manager.

A call-answering mode in which an agent automatically receives ACD calls without pressing any button to receive calls.

Agents enter the Auxiliary (AUX) work mode for non-ACD activities, such as taking a break, going for lunch, or making an outgoing call. Agents in the AUX work mode are unavailable to receive ACD calls.



35

Glossary

Avaya Aura ® bridged appearance

CMS

CWC

EAS

IMS manual-in

MAO network region node principal private network public network

A converged communications platform unifying media, modes, network, devices, applications. Avaya Aura ® is based on the SIP architecture with

Session Manager at the core.

A call appearance on a telephone that matches a call appearance on another telephone for the duration of a call.

A software program for reporting and managing agents, splits, trunks, trunk groups, vectors, and VDNs. With Call Management System (CMS), you can also administer some ACD features.

Call Work Codes (CWCs) are up to 16–digit sequences that agents type to record the occurrence of customer-defined events, such as account codes or social security numbers.

A feature that Communication Manager uses to distribute calls based on agent skills. With Expert Agent Selection (EAS), you can ensure that callers connect to agents with the required skills.

IP Multimedia Subsystem (IMS) is an architectural framework for delivering IP multimedia services.

A call-answering mode in which an agent must press manual-in to receive an ACD call.

Maximum Agent Occupancy (MAO) is a feature that Communication

Manager uses to set thresholds on the amount of time that an agent spends on a call. The MAO threshold is a system-administered value that places an agent in the AUX work mode when the agent exceeds the MAO threshold for calls.

A group of IP endpoints and Communication Manager IP interfaces that are interconnected by an IP network.

A network element that connects more than one link and routes voice or data from one link to another. Nodes are either tandem or terminal.

Tandem nodes receive and pass signals. Terminal nodes originate a transmission path or terminate a transmission path. A node is also known as a switching system.

A terminal that has the primary extension bridged on other terminals.

A network used exclusively for the telecommunications needs of a particular customer.

A network that can be openly accessed by all customers for local and long-distance calling.



36

SIP trunk trunk allocation work mode

SIP

Session Initiation Protocol (SIP) is an application-layer control signaling protocol for creating, modifying, and terminating sessions with more than one participant using http like text messages.

A dedicated telecommunications channel between two communications systems or Central Offices (COs).

The manner in which trunks are selected to form wideband channels.

A function that an agent performs during the work shift. ACD work modes include AUX work, auto-in, manual-in, and ACW.



37

Index

Numerics

911 calls ...............................................................................

17

96X1 SIP phones .................................................................

21

active VDN ....................................................................

22

feature support .............................................................

22

UUI ...............................................................................

22

versus H.323 phones ....................................................

21

A

active VDN ...........................................................................

22

adding stations .....................................................................

19

adding users ........................................................................

19

administration tips ................................................................

11

after call work (ACW) ...........................................................

23

automatic alternate routing (AAR) .......................................

10

automatic route selection (ARS) ..........................................

10

auxiliary (AUX) .....................................................................

23

Avaya support website support ............................................

34

E

embedded templates ............................................................

11

event packages ....................................................................

25

evolution server ..........................................................

7 ,

10

,

11

F

feature access codes (FAC) ................................................

10

feature buttons .....................................................................

18

after-call ........................................................................

25

auto-in ...........................................................................

25

auto-msg-wt ..................................................................

25

aux-work .......................................................................

25

coach ............................................................................

25

manual-in ......................................................................

25

q-calls ...........................................................................

25

stroke-cnt ......................................................................

25

uui-info ..........................................................................

25

vu-display .....................................................................

25

work-code .....................................................................

25

feature invocation failure ......................................................

11

feature server .......................................................................

11

full-call model .......................................................................

10

B

backup files ..........................................................................

21

button-based deskphones ...................................................

21

button modules ....................................................................

21

C

call handling time .................................................................

16

call queuing time ..................................................................

16

call traffic and signaling, separating .....................................

11

CM screens

IP Network Region ........................................................

14

IP Node Names ............................................................

14

Numbering-Private Format ...........................................

18

Off-PBX-Telephone Station-Mapping ............................

19

Signaling Group ............................................................

15

Station ..........................................................................

18

Trunk Group ..................................................................

16

CM templates .......................................................................

11

communication profile ..........................................................

19

concurrent SIP agents .........................................................

25

configuring IP nodes ............................................................

14

D

dedicated SIP trunk groups ..................................................

11

denial events ........................................................................

27

I

implementation checklist ........................................................

7

INVITE message ..................................................................

16

IP-IP direct audio .................................................................

14

IP node names, administering .............................................

10

L

line side ................................................................................

21

logical agent .........................................................................

27

login .....................................................................................

23

logout ...................................................................................

23

M

media encryption ..................................................................

11

message

INVITE ..........................................................................

16

NOTIFY ........................................................................

25

PUBLISH ......................................................................

25

SUBSCRIBE .................................................................

25

multiple call handling (MCH) ................................................

27

N

NOTIFY message ................................................................

25



38

Index numbering plan, administering .............................................

10

O

off-pbx station (OPS) ............................................................

11

off-pbx telephone integration and mobility (OPTIM) .............

11

OPTIM OPS signaling ..........................................................

11

P

parameters

APPSTAT ......................................................................

21

DSCPSIG .....................................................................

21

DSCQUAD ....................................................................

21

ENABLE_CALL_LOG ...................................................

21

ENABLE_CONTACTS ..................................................

21

ENABLE_MODIFY_CONTACTS ..................................

21

ENABLE_REDIAL ........................................................

21

L2QSIG .........................................................................

21

L2QUAD .......................................................................

21

OPSTAT ........................................................................

21

PROVIDE_LOGOUT ....................................................

21

Personal Profile Manager (PPM) .........................................

25

processor ethernet (PE) .......................................................

11

public safety answering point (PSAP) ..................................

17

PUBLISH message ..............................................................

25

SIP phone buttons (continued) call-appr ........................................................................

22

manual-in ......................................................................

22

q-calls ...........................................................................

22

stroke-cnt ......................................................................

22

uui-info ..........................................................................

22

vu-display .....................................................................

22

work-code .....................................................................

22

SIP phone scalability ...........................................................

25

SIP settings ..........................................................................

11

SIP traffic .............................................................................

16

SIP URI conversion .............................................................

14

site-specific option number (SSON) ......................................

7

SRTP considerations ...........................................................

11

station mapping mode .........................................................

19

stations, adding ....................................................................

19

SUBSCRIBE message ........................................................

25

support .................................................................................

34

survivability ............................................................................

7

survivable core .....................................................................

11

survivable remote .................................................................

11

system failover .......................................................................

7

system management interface (SMI) .....................................

7

Q

quality of service (QOS) .......................................................

21

R

real-time transport control protocol (RTCP) .........................

14

reason code .........................................................................

23

redirect on OPTIM failure (ROOF) .......................................

16

route pattern, administering .................................................

10

route pattern selection .........................................................

17

T

time server settings ................................................................

7

touch-based deskphones .....................................................

21

trace command ....................................................................

27

tracing SIP messages ..........................................................

29

transmission control protocol (TCP) .....................................

11

transport layer security (TLS) ...............................................

11

troubleshooting

96X1 SIP agent deskphones ........................................

27

Communication Manager 6.2 .......................................

27

scenarios ......................................................................

29

Session Manager ..........................................................

29

trunk groups, administering .................................................

10

trunk side .............................................................................

21

trust management ..................................................................

7

S

secure real-time transport protocol (SRTP) .........................

11

separating call traffic and signaling ......................................

11

servers .................................................................................

11

service observing .................................................................

22

session refresh interval ........................................................

16

simple network time protocol (SNTP) ...................................

11

SIP agent login ....................................................................

22

SIP agent logout ..................................................................

22

SIP phone buttons after-call ........................................................................

22

agnt-login ......................................................................

22

agnt-logout ....................................................................

22

aut-msg-wt ....................................................................

22

auto-in ...........................................................................

22

aux-work .......................................................................

22

U

uniform dial plan, administering ...........................................

10

uniform dial plan (UDP) .......................................................

10

users, adding .......................................................................

19

V

videos ..................................................................................

33

W

work mode buttons ..............................................................

18

work mode changes .......................................................

23

,

24



39

No results