ACS Integration with

Solidus eCare

INSTALLATION INSTRUCTIONS

© 2014 Aastra Technologies Limited. All rights reserved.

This document contains proprietary information, which is protected by copyright. No part of this document may be reproduced or transmitted in any form or by any means, electronic or mechanical, including photocopying, recording, or by any information storage and retrieval system, or translated into another language, without prior written consent of Aastra Technologies Limited, Concord, Ontario, Canada. NOTICE

The information in this document is subject to change without notice. AASTRA MAKES NO WARRANTY OF ANY KIND WITH REGARD TO THIS MATERIAL, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. AASTRA shall not be liable for errors contained herein nor for incidental or consequential damages in connection with the furnishing, performance or use of this material. Aastra Technologies Ltd. Concord, Ontario, Canada

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1 Introduction This document describes installation and integration of Aastra Connectivity Server (ACS) with Solidus eCare. ACS provides call control connectivity to Call Managers supporting the SIP protocol. For this configuration, ACS is used instead of Open Application Server (OAS) for call and media control. Call Managers supported for this configuration include the following:

Aastra 400

Cisco

Aastra MX-ONE *Note: MX-ONE can be connected to ACS for mixed Call Manager environments, such as MX-ONE and Lync

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2 System Architecture Figure 1 represents an overview of the system architecture when Solidus eCare is integrated with ACS.

Figure 1: Overview of the system components

Telephony Application Server (TAS) and SIP Proxy/Registrar These components are contained with the same Windows service. TAS exposes a CSTA Phase III interface to connected CSTA clients, to make telephony requests and receive events. The SIP Proxy/Registrar provides an endpoint for SIP clients to register toward as well as to receive and send SIP events.

Media Server The CMG Media Server provides media integration, such as playing messages, collecting DTMF digits, and conferencing multiple parties together.

BSA BluStar Agents running as SIP Soft clients register as SIP clients toward the SIP Registrar in the TAS service.

Solidus eCare

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The Solidus eCare Call Control Service connects to the CSTA interface in TAS to send call and media requests and receive events. TAS provides a call control and media interface to the Solidus eCare Call Control Service. Other Solidus eCare services requiring call and media control, such as the Router Service, Agent Service and Script Manager, connect to the Call Control Service.

Call Manager The Call Manager supports SIP trunks which are configured to route into TAS. This allows incoming service group calls to be routed to BSA agents via TAS. In addition, the Call Manager can configure BSA extensions to route into TAS so that agents’ extensions may be dialed directly from the other extensions in the Call Manager.

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3 Installation Instructions Following are the steps required to install Solidus eCare with ACS.

Install Media Server

Install TAS

Configuration of Media Server and TAS Components

Install Solidus eCare

Configure Call Manager data

Configure Call Manager SIP trunks

Each step is detailed below.

The TAS, Media Server and Solidus eCare installations can be launched from the Aastra Package Browser.

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3.1 Media Server Installation NOTE: Media Server Installation is done silently if done from the package browser.

If the setup is launched from the .msi package on the installation media the following dialogs will be seen.

Next, a license acknowledgement screen is seen.

The installation will default to using localsystem for the service account or a domain account can be specified.

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Next is the program files destination dialog.

On the next screen, click Install.

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After files are copied the installation completes.

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3.2 TAS Installation

NOTE: TAS Installation is done silently if done from the package browser.

If the setup is launched from the .msi package on the installation media the following dialogs will be seen.

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Once installed, TAS and the Media Server can be stopped and started from the Services control panel

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3.3 TAS and Media Server Configuration After the Media Server and TAS are installed, there are configurations that need to be made. The TAS and Media Server configuration is done by way of a small GUI that is available from the Start menu after it has been installed.

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After installation the Media Server configuration can be left at default (except for the AudioFiles Prefix setting) unless otherwise conflicting with ports in use. An example Media Server configuration is seen above: NOTE: Be sure to take note of the backslash after the Audio Files Prefix path. This is necessary because of the need to now use a “set default container” block in Scripts and the leading backslash is not accepted in that block. Be sure to test router SA’s with this change to make sure the prompts associated with those are played OK. An example of how the prompt files and folders might be seen on the disk is shown following:

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3.3.1 TAS and Media Server Configuration Data Location The TAS configuration is stored in the registry at HKLM\Software\Aastra\Tas.

The Media Server configuration data is stored in the registry at HKLM\Software\Aastra\Mediaserver. An example of the registry keys are seen following:

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3.3.2 Media Server Settings

The settings are described in the table below.

Setting Description Default Value

SIP port The port TAS will use to connect to the Media Server. :port for the default Ethernet interface, <interface>:port for a specific Ethernet interface.

:5065

Dialog TTL The interval for a simple “session timer” that uses OPTIONS SIP message to periodically check if the call is still up

10 mins.

RTP port range

Ports used for RTP 40000-50000

RTCP If the media server sends RTCP Sender Reports and Receiver Reports

Unchecked

MOH file File to be used for music on hold feature (Solidus does not use this)

<InstallDir>\Ringing.wav

Trim Recordings

If the media server should remove trailing silence from recordings

Checked

Codec Preference

Lists the codecs, in order of preference

pcma, pcmu, g.722, g.729, rfc2833

Audio Files Prefix

The location where the message prompt files are located.

C:\temp\

SRTP SDP Offer

If the media server will include a crypto attribute (as described in RFC 4568) in the SDP offer generated

Unchecked

SRTP Best Effort

If this is not checked then “Strict SRTP” is used. With this checked the SDP offer has the profile RTP/AVP while strict means that the profile RTP/SAVP will be used. Best effort allows for SRTP to be

Unchecked

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turned off through SIP negotiation while strict will not allow SRTP to be turned off if the call has started with SRTP. ** See note below

Default Recording Rate

8kHz or 16kHz 16kHz

Log Path The location where the Media Server log files are located

<InstallDir>\Logs

Log Level Indicates the amount of detail to be logged.

Trace

Max size Maximum size of the log file. After the maximum size is reached a new log is opened. Size is unrestricted if left at default and a new log is opened each day

0

Discard after The amount of time the log file will be maintained before it is discarded.

7 days

** Please note that for SRTP there is no way to force it one way or the other. If an INVITE with (or without) crypto attribute is received then the media server will always answer with (or without) crypto attribute regardless of the settings. For calls without SDP, SRTP will be enforced if SRTP SDP Offer is selected and SRTP Best Effort is not selected. If any configuration changes are made other than the log level, the Media Server does need to be restarted.

3.4 TAS Configuration Details TAS configuration items are seen below:

Setting Description Default Value

Registrar SIP Listening Port

“Listening Port”. TAS will listen on this port for both TCP and UDP

5060

CSTA3 Service Listening Port

This is the port the TAS CSTA3 Web Service is listening to. By default only TCP bindings are used.

8732

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Media Server Always In Call

Used for Cisco integration, required to be checked when using Cisco

Unchecked

Default Route The IP address of the Call Manager. ;lr specifies “loose routing” and is recommended for call managers that support it

sip:127.0.0.1:5060;lr,transport=TCP

Default Domain The domain used for TAS originated calls. (TAS just uses numbers so if a call to an agent (e.g. 30000) is placed it will be placed to sip:30000@defaultdomain

aastra.com

Media Server Uri Uri for Media Server host sip:127.0.0.1:5065,transport=TCP

BVD Number Ranges

Number series used by TAS to identify monitored entities as Basic Virtual Devices for SeC Service Access configuration

40000-40009, 80000-80009

Agent Number Ranges

Number series used by TAS to identify monitored entities as extensions. Ranges and single numbers are accepted, i.e. 1, 2, 3-12, 28-99, 100, 200-300, etc.

30000-30009

LogPath Path to TAS logs C:\logs

LogLevel Verboseness of TAS log 9

3.5 Install Solidus eCare

During the installation of Solidus eCare 8.3 there is a new Call Manager Integration option to use either an OAS or an ACS based system. When prompted for the Call Manager type, select Aastra Connectivity Server (ACS). It is not necessary to install OAS.

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It is also possible to modify the Call Manager type from Solidus eCare Setup as shown below.

Changing the Call Manager will require a restart of the Solidus eCare services.

3.6 Configure SeC Call Manager Data

When Solidus eCare is integrated with ACS, Call Manager data must be configured in Solidus eCare Configuration Manager. Below is the data that must be configured.

3.6.1 Message Prompt Files

Message prompt files should be copied from the Solidus eCare installation DVD to the location of the Media Server. The files should be copied into the directory specified as the Audio Files Prefix in the Media Server Configuration. For example, if c:\voice\system is configured as the Audio Files Prefix, the files should be copied into this directory under a subdirectory

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indicating the supported language. For example: c:\voice\system\USEnglish. The system message prompt files that traditionally were installed by an OAS installation are now available from the Solidus eCare installation media. If custom message files are defined, they should be added to the appropriate directory depending on the language. For example, if custom message files are used in a non-Script Manager Service Access or in a Service Group defined with the language USEnglish, the files should be copied to the c:\voice\system\USEnglish directory so that they will be accessible to the system.

3.6.2 ACS Site Setup

From Configuration Manager Advanced System Properties, add a site for the ACS Server.

Figure 2: ACS Site Configuration

Currently, only one ACS Call Manager server is supported for an ACS site. After adding the site, add the ACS Call Manager Server. The Server Location should indicate the machine where the TAS Service is installed. The Server Port should match the port configured for CSTA connections on the TAS Service.

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Figure 3: ACS Site Configuration – SIP Softphone Settings On the IP Softphone tab, configure the SIP Settings. The Proxy field should be set to the IP address or hostname of the machine where the TAS Service is installed. The Proxy port should be set to the same port that is configured as the Registrar SIP Port on the TAS Server. For ACS integration do not configure any Access Code digits for Conference or Transfer. If present this will cause BSA to attempt transfer using these access code digits instead of a SIP REFER method for conference and transfer. For information on other settings, consult the Configuration Manager Online Help. After the ACS site is added, Configuration Manager will display it under the Call Manager Resources folder as shown below in Figure 3.

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Call Manager data including BVDs, Languages and Play Messages can now be configured.

3.6.3 BVD Configuration

Add the Basic Virtual Devices, or BVDs, which will be used to route service group calls to Solidus eCare from the Call Manager. Each BVD number should correspond to the SIP trunk configured to route from the Call Manager to TAS (refer to the ACS Configuration tool for BVD Number Ranges).

3.6.4 Play Message Lists

Play Messages such as queue welcome messages and repeat queue messages, are grouped together into a Play Message List. If your contact center is using multiple languages, it is recommended to add a separate Play Message for each language. At least one Play Message List must be defined for the system.

3.6.5 Languages and Play Message Lists

The Languages and Play Message Lists are unique among tenants. There are no common languages or play message lists. If for example two tenants on a particular system wish to use the Spanish language, a unique Spanish language would need to be created for each tenant. The .rul files and the various subfolders for the “root container” are common between all tenants

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on the system and the relative path of the language specified media files are passed to the media server.

3.6.5.1 Play Messages

After the Play Message List is defined, Play Messages can be added to the list. The message prompt files provided with Solidus eCare can be utilized in the Play Messages, or it is possible to record new message files and use those. On the General tab of the Play Message Properties, enter the Identification number for the Play Message. This number should be unique within the Play Message List. Enter a description for the Play Message to help you identify its meaning. The Media Objects tab can be used to configure the content of the Play Message. For details on the various options available, consult the Play Messages User Guide (4-1553 FAS10455).

3.6.5.2 Languages

After the Play Message List and Play Messages are defined, a Language can be configured. At least one Language is required.

Any descriptive name can be entered in the Name field. The language Rule File designates the rules for how particular values, such as numbers, dates and time will be formatted. Enter the name of the rule file that should be applied for the language being defined. Rule files are installed on the Solidus eCare Server at <InstallDir>\Services\Rule. The path does not need to be provided in the Rule File field, only the name of the rule file to be used. In the Prompt Path, enter the relative location of the message prompt files to be used for this language. The value entered will be appended to the value defined as the Audio Files Prefix for the Media Server. In the example above, if the Audio Files Prefix is defined as c:\Voice\Files, the prompts for

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this language would be expected to be located at c:\Voice\Files\swedish. Ensure that the files exist at that location on the Media Server. From the Play List drop-down list, select the Play Message List that will be used by this language. Once the Call Manager data is defined, including the ACS Site, BVDs, Play Messages and Languages, other Solidus eCare data can be configured, including Service Groups, Service Accesses, Agent Groups, and Agents. For details on configuring this data, consult the Configuration Manager Online Help.

4 Solidus eCare/ACS Call Manager SIP Trunk Conformance

Once a SeC/TAS based system has been installed and configured for a particular call manager, the following set of test cases should be performed to verify the functionality of the system. For this set of test cases, an ACS Site is defined with one TAS based Solidus eCare system. DM, CCAS, ROS and SM AppMediaService communicate with TAS through the CallControlServiceLink.dll, which then sends the request to the Solidus Call Control Service. These test cases are designed to test call manager interaction with an ACS site, as well as call and media control through the ACS interface. The target call manager shall have at least one SIP trunk configured towards the Solidus/ACS system. The access numbers for the trunk will be used in the configuration of Solidus eCare service access’ and system requeue device. The access numbers for the SIP trunk applications as well as the agent device extension numbers will be defined and configured in the ACS Configuration tool on the SeC/ACS system.

4.1 TAS Service Start/Restart/Monitoring Test Cases

4.1.1 Service Access Monitor Start and Stop from SeC Configure a SeC SA using the SIP trunk access number as the number to monitor. Confirm in the TAS log that a monitor can be started when the SA is activated with a unique monitor cross reference ID generated. Confirm

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that this unique monitor cross reference ID is used to stop the monitor on the configured SA device when the SA is deactivated.

4.1.2 Service Access Monitor Start and Stop from Call Manager

Disrupt the SIP trunk connection from the call manager to SeC and confirm that the monitor is stopped and that it is restarted when the SIP connection from the call manager is re-established.

4.1.3 Requeue Device Monitor Start and Stop Configure a requeue device in CM Contact Center properties on the Call tab using the access number of a SIP trunk coming from the target call manager. Confirm in the TAS log that a monitor is started on the device.

4.1.4 Restart Call Control Service With connected SIP trunks, restart the Call Control service. Verify that the SA’s lose the monitors and that upon restart of the Call Control service that the SA monitors are successfully restarted.

4.2 Call Control Functionality Test Cases

BSA is using softphone connected to ACS in all the cases below.

4.2.1 Start BSA with Softphone

Verify BSA starts up properly, and the extension can be monitored via ACS.

4.2.2 Queued Call

Place an incoming call to the SA via the SIP trunk access number and have it queue for a Service Group such that repeat queue messages defined for the group are heard repeatedly.

4.2.3 Incoming Call via Router Service Access

Place an incoming call to a Router SA via the SIP trunk access number and have the call routed to BSA and answer. Verify that the agent enters Talking state and that an audio path is established between the caller and the BSA agent.

4.2.4 Incoming Call via Script Manager Service Access

Place an incoming call to a Script Manager SA via the SIP trunk access number and have the call routed to BSA and answer. Verify that the agent

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enters Talking state and that an audio path is established between the caller and the BSA agent.

4.2.5 Outgoing Call

Place an outbound call on the SIP trunk from BSA and answer at the far end. Verify that the agent enters Talking state and that an audio path is established between the BSA agent and the external endpoint.

4.2.6 Hold/Retrieve

Hold and retrieve incoming and outgoing SIP trunk calls call between BSA and the external endpoint. Verify the state is correct and that the audio path is correct for each state.

4.2.7 Clear Call

Clear a call from BSA in various states: Calling, Talking, Conference. Verify that the call is removed from the SA and is seen as terminated from the perspective of the external call manager.

4.2.8 Consultation Call

With an existing call in Talking state, place a new call over the SIP trunk to an endpoint on the external call manager. Verify that the new call can be answered and displays in Talking state, while the original call is in Held state.

4.2.9 Transfer Incoming Call

From BSA, transfer an existing incoming SIP trunk call to another BSA agent. Repeat transferring before answer and after answer by the other BSA agent. Verify that original BSA is idle when the transfer is complete and that the audio path is established correctly between the other BSA agent and the incoming SIP trunk caller.

4.2.10 Transfer Outgoing Call

From BSA make an outbound call over the SIP trunk to an extension on the call manager that is not monitored by TAS. This will be a call using the default route defined for TAS. Once this outbound call is established and speech path is confirmed, transfer the call to another BSA agent (repeat for both announced and blind transfers) and again confirm speech path with the connected parties. Verify that the call is torn down properly regardless of whether the BSA agent or the external caller disconnect first.

4.2.11 Conference Incoming Call

Receive an incoming SIP trunk call by BSA and then create a conference between BSA, the incoming SIP trunk caller and another extension defined

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on the call manager. Verify speech path and call window state display is correct for all parties. Clear BSA from the conference and verify that the BSA call window shows as Idle and that the audio path is maintained between the two remaining parties. Clear the call between the two remaining parties.

4.2.12 Conference Outgoing Call

Create an outgoing SIP trunk call by BSA and then create a conference between BSA, the outbound call and another extension defined on the call manager. Verify speech path and call window state display is correct for all parties.

Clear BSA from the conference and verify that the BSA call window shows as Idle and that the audio path is maintained between the two remaining parties. Clear the call between the two remaining parties.

4.2.13 Deflect Call

From BSA, deflect an incoming SIP trunk call to another extension or destination defined on the call manager. Deflect from both ringing state and connected state.

4.2.14 DTMF Digits

From BSA, enter DTMF digits for an existing call. Verify that the digits are sent to the opposite party. Various combinations of play message interruption by digits, inter digit timeout, termination digit, and flush buffer options in the Script Manager Collect Digits block are to be verified.

4.2.15 After-Call Handling

Configure After-Call handling for a service group, and configure to send the agent ID with the call. Place an incoming SIP trunk call to BSA and then send the call to the after-call handling destination. Confirm that the call is properly deflected, that correct call window states and displays are seen and that correct audio path is established between the SIP trunk caller and the After Agent Handing destination.

4.2.16 Deflect to Service Group

Deflect an incoming SIP trunk call to another service group. Verify that the call is correctly deflected and routed through the service group.

4.2.17 Reject Service Group Call

Reject an incoming service group call, and verify that the call routes to the requeue destination, and it is routed to another agent. Repeat allowing the call to timeout and be handled by the requeue device.

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4.2.18 Associate Data

Use a Script Manager Service Access and associated Script Designer script that utilizes an “Associate Data” block and configure the contents of the block to be a maximum length string of 512 digits (this block in Script Designer is limited to 100 characters). Confirm that the data is tagged to the call and displayed on another BSA when the call is transferred to another agent.

4.2.19 Assist

From a BSA agent, request Assist from another agent. Verify that the assisting agent is able to intrude on the call properly and the state display is correct on both agents, during the assist as well as after the assisting agent disconnects and when the incoming caller disconnects.

4.2.20 Single Call Monitoring

From a BSA agent, request to Monitor another BSA agent for a single call. Verify that the monitoring agent is able to intrude on the call properly and the state display is correct on both agents.

4.2.21 Continuous Call Monitoring

From a BSA agent, request to Continuously Monitor another BSA agent.. Verify that the monitoring agent is able to intrude on the first call properly as well as all subsequent calls and the state display is correct on both agents.

4.2.22 Callback Handling

Configure a service group to ask for callbacks, and add a call to the queue that is changed to a callback. Verify that the agent is prompted to make the callback, and the callback can be initiated correctly from BSA.

4.2.23 Web Callback Handling

Add web callbacks to the system. Verify that the agent is prompted to make the web callback, and the callback can be initiated correctly from BSA.

4.2.24 Campaign Call Handling Verify that campaign calls (regular and progressive) can be handled by BSA agents.

4.2.25 Dispatch Call Handling

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Verify that an incoming call can be directed to a dispatch SG and that the call can be retrieved from the dispatch window.

4.2.26 Common Hold Call Handling

Verify that a call can be placed on common hold and can then be retrieved.

4.2.27 BSA Call Recording

Verify that the “Record Calls” feature in BSA can be initiated and calls that were recorded can be played back successfully.

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5 CISCO SIP Trunk Characteristics

5.1 SIP Trunk Characteristics for Cisco UCM

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5.2 SIP Trunk Security Profile

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5.3 SIP Profile

Same page but further down…

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5.4 SIP Trunk configuraion

same page

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Same page but further down…

Same page but further down… Note: Be sure to match the destination port configured as seen below with the TAS SIP Listening Port configured on the ACS/SeC system. This configuration on the ACS/SeC system is done with the ACS Configuration Tool.

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5.5 Route numbers to the SIP trunk

Note: Be sure to set the Route Pattern as seen below to correspond to what is

configured for the BVD range in the ACS Configuration tool on the ACS/SeC system.

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ACS Configuraton Tool For CISCO Call Managers it is necessary to configure the TAS service to always have the Media Server in the call.

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6 Characteristics of the Aastra A400 Integration The following information describes the SIP Trunk configuration and some maintenance tools for the Aastra A400.

6.1 Aastra 400 Web Administration Tool

The A470 system is configured via a web interface. Open a browser towards the IP address of the system and login (admin, xxxxxxx)

The SIP trunk from the A470 should be configured as “SIP Provider” as opposed to “SIP Networking”.

6.1.1 Characteristics of the A470 Network Interface (SIP Trunk)

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6.1.2 Characteristics of the A470 Trunk Group

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6.1.3 Characteristics of the A470 ROUTE

6.1.4 Aastra 400 Series Maintenance Monitor

Access to the A470 Maintenance Monitor is via telnet, port 1818. Prompts are seen for Username and Password (user: admin, pwd:xxxxxx). This maintenance monitor can be used to collect diagnostic and trace information. Rolling predefined traces are defined for basic troubleshooting. Once logged on to the maintenance monitor the menu looks as follows:

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The trace output can be specified to go to the screen or to a file. Under the rolling trace menu there is a toggle for trace output destination.

When the output is to screen the contents of the telnet window can be copied to the clipboard and when the destination is specified as file, the output is written to a log on the A470 server file system. Once the output destination is set, start the pre-defined rolling trace.

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When the destination output is set to file, after the trace activity has been done, stop the rolling trace and locate the file via ftp to the A470. FTP access is via a browser to the IP address of the A470 server. Once logged on, browse to the logs folder and see mmtrace.txt

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Open the file and copy the contents

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7 Limitations

Only BluStar Agent SIP Softphone is supported for agent applications. Desktop Manager, hard phones, and Phone Agents are not supported for the ACS solution.

One TAS Server can be installed in the system. Multiple TAS servers are not supported.

ASR and TTS are not supported with the ACS solution.

ACS and OAS cannot be run simultaneously.

Tone Generator resources are not supported with the ACS solution.