ACS Integration with

Solidus eCare
INSTALLATION INSTRUCTIONS

 2014 Aastra Technologies Limited. All rights reserved.

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consent of Aastra Technologies Limited, Concord, Ontario, Canada.
NOTICE
The information in this document is subject to change without notice.
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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

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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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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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 SAs 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
SIP port

Dialog TTL

RTP port
range
RTCP

MOH file

Trim
Recordings

Codec
Preference
Audio Files
Prefix
SRTP SDP
Offer

SRTP Best
Effort

Description
The port TAS will use
to connect to the
Media Server. :port
for the default Ethernet
interface,
<interface>:port for a
specific Ethernet
interface.
The interval for a
simple session timer
that uses OPTIONS
SIP message to
periodically check if
the call is still up
Ports used for RTP

Default Value
:5065

If the media server

sends RTCP Sender
Reports and Receiver
Reports
File to be used for
music on hold feature
(Solidus does not use
this)
If the media server
should remove trailing
silence from
recordings
Lists the codecs, in
order of preference
The location where the
message prompt files
are located.
If the media server will
include a crypto
attribute (as described
in RFC 4568) in the
SDP offer generated
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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10 mins.

40000-50000

<InstallDir>\Ringing.wav

Checked

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

C:\temp\

Unchecked

Unchecked

16

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
8kHz or 16kHz

Default
Recording
Rate
Log Path

Log Level
Max size

Discard after

The location where the

Media Server log files
are located
Indicates the amount
of detail to be logged.
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
The amount of time
the log file will be
maintained before it is
discarded.

16kHz

<InstallDir>\Logs

Trace
0

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
Registrar SIP
Listening Port

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

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

Unchecked

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

BVD Number
Ranges

Uri for Media Server host

Number series used by TAS to
identify monitored entities as
Basic Virtual Devices for SeC
Service Access configuration
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.
Path to TAS logs
Verboseness of TAS log

sip:127.0.0.1:5065,transport=TCP
40000-40009, 80000-80009

Default Route

Agent Number
Ranges

LogPath
LogLevel

3.5

sip:127.0.0.1:5060;lr,transport=TCP

30000-30009

C:\logs
9

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.

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
SAs 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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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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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 predefined 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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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.

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