Installation and administration¶

Requirements¶

Workstation

DesktopClient is a thick desktop application for Windows built on the .NET and WPF technologies. Therefore, it does not pose any special workstation requirements.

Network settings for ProServer

DesktopClient requires the possibility to connect to ProServer. You can set the location of the ProServer and the port (the default for TCP is 18321) with the PbxParameters configuration parameter.

Network settings for softphone

If you are going to use the embedded softphone, you must enable SIP communication with the server. See Network communication.

Installation¶

The DesktopClient application itself does not require installation. Copy the application build folder to the target computer, then just run iCC.DesktopClient.exe.

You must install the Microsoft Edge WebView add-on for the chat elements of the application to function properly. It allows the application to natively open a web interface that provides ReactClient functionality.

Navigate through the link to the add-on website
Choose the appropriate installer according to the operating system architecture
Restart your computer after the installation

Note

It is advisable to keep this add-on up to date due to security threats. On the other hand, it is always advisable to test the functionality with a given Desktop client build on one specific computer before deploying a new version on all computers. After a successful test, then deploy to the remaining ones.

System settings for application integration¶

Attention

When integrating the application, it is necessary to modify the DeletePerso'' script on the installation so that it does not delete the data related to the DesktopClient. The ``Perso table contains saved application bar definitions and reaction definitions (each agent has its own definitions). The database RefName for these definitions is ProCaller#RibbonDefinition and ProCaller#EventHandlingDefinition.

Steps required to integrate the application:

ProServer settings
Definitions for the database
Local configuration file
Login
Workplace
LiteralLookup set
Network communication
Basis for user statistics
Default sound device (WebRTC only)

ProServer settings¶

DesktopClient does not communicate with FrontStage directly, but via ProServer. The station must therefore have network access to ProServer and have DesktopClient set appropriately.

The username and password are not required since the application uses Windows authentication and the current user is sent to ProServer. For workstations connected to a domain, these are usernames in the form domain\name (e.g. acmecorp\johndoe).

In the ProServer administration in the Agents section, create an account necessary for the connection as follows:

In the Agents section, name the account in the Name field with the name of the agent or the like.
In the Credentials section, add or edit the entry as follows:
- Enter the username in the System Name field as it is sent by Windows authentication from stations (e.g. acmecorp\johndoe) (see above).
- In the Application field, select iCC.DesktopClient
In the Permissions section, add the necessary roles. Usually UseChannels a ExtensionCTI.
In the Data Items section, add or modify the entry as follows:
- Enter UserData in the Key field
- In the Application field, select iCC.DesktopClient
- In the Data field, enter a string in the style of <?xml version="1.0" encoding="utf-8"?><UserData><IdentityCC>agent login< /IdentityCC></UserData>, e.g.
```
<?xml version="1.0" encoding="utf-8"?><UserData><IdentityCC>fs-sql_demo\jelinek</IdentityCC></UserData>
```
  See System Name.

Tip

In the Stations section, you can track several recent DesktopClient connections.

Definitions for the database¶

In the Perso database table, it is necessary to upload the bar definition and the reaction definition for the users who are to use the application. The definitions are saved in the json format.

Database entries and their RefName:

Ribbon definition - ProCaller#RibbonDefinition
Response definition - ProCaller#EventHandlingDefinition

Local configuration file¶

The configuration parameters for the connection are set for each computer separately. The file name is appsettings.json; it is located in the application folder. A local settings in the config file has lower priority than workplace settings in the database, if both places contain the same parameter, the database one will be used.

WebRTC¶

For locally usable parameters see WebRtc Section. At least ServerUrl parameter is mandatory.

SIP¶

The following information needs to be filled in:

PbxParameters
InstanceName
SipServer and SipServerPort

Login¶

Agent has to be able to login using the application. You can set this via Agent visual editor.

Workplace¶

Workplaces which are supposed to work with the DC app have to log-in to the ProServer as a softphone. In ReactAdmin, Agents ‣ Workplace section, Softphone configuration (JSON) field, set the json softphone object. The order of the parameters in the object does not matter.

WebRTC¶

To be able to log-in from DC to AS7 it is necessary that PhoneType parameter is in a format PhoneName_DC (connection won’t work without the “_DC” stated)

An object example:

{"PhoneType": "AS7anCTI_DC", "Password": "password", "UserName":"user@user.com", "ServerUrl": "https://server.atl.com/cti/ancti/ancti.js", "DeviceName": "sip:555@stat.777", "HoldOnDirectCti": false}

For possible parameters and their meaning see WebRtc Section.

SIP¶

An object example:

{"PhoneType": "SWPSIP", "SipServer": "192.168.1.221", "Password": "xyz", "UserName":"420"}

The object will also include other attributes with the SIP server location, verification, etc. (see Configuration).

LiteralLookup set¶

For the correct rendering of all texts in the application, it is a good idea to always upload an up-to-date set of LiteralLookup texts to the database. This will limit the situation where only the system LiteralText is displayed in the application, without the corresponding text, for the given language.

Network communication¶

WebRTC¶

Communication via internet, see Network communication

SIP¶

Enable network communication between the station and the PBX specified in Softphone configuration (JSON). It is normally handled by the FrontStage network specialist, see Network communication.

Basis for user statistics¶

If you want to fill statistics for individual users, you need to set the following:

BroadcastDataQueries - in the administration, set the parameter to true (if you set it on an already “live” installation, you must then restart the synchronous service - the parameter is loaded at start-up)
Create a database query (Administration, section Data queries) that returns the required columns for statistics
- The Validity interval item must be set for the query
- The query is always called after the specified interval; it is evaluated as a whole and the data is distributed to individual agents in their applications
- It must contain the AgentId control column to allow data binding
You must assign the created database query to a specific agent
- In the administration in the Agents section, fill in the DQ in the bar parameter

Note

After creating a new database query, check that the indexes are set correctly. This will prevent complications due to increased computing power on the server side.

Default sound device (WebRTC only)¶

See Default communication device.

Configuration file¶

The settings are defined in the appsettings.json configuration file, which is located in the application folder.

Important

The configuration file is extensive. Do not edit sections outside this description - they are not meant to be configured by users. Make sure to follow the syntax of the JSON file, such as nesting elements, enclosing attributes in quotes, etc.

Configuration file:

{
"$type": "OnPremiseStrategy",
"DiscoveryHost": "https://localhost:5001/",
"Smtp": {
   "ServerName": "127.0.0.1",
   "ServerPort": 1025,
   "UserName": "atlantis@krch.tech",
   "Password": "DqyMlwqoNIWikzC9YwVUEw",
   "From": "atlantis@krch.tech",
   "SslMode": "None"
},
"DesktopClientConfiguration": {
   "RunOnboardingRoutine": true,
   "ReactClientOnboardingUrl": "https://localhost:5001/api/onboarding/accept/dc/{onboardingKey}",
   "ReactClientFileResourceUrl": null,
   "ReactClientTextResourceUrl": null,
   "Debug": false,
   "DebugIndicators": "App,Detail,Handling",
   "Logging": {
      "LogLevel": {
      "Default": "Trace",
      "System": "Information",
      "Microsoft": "Error",
      "iCC.DesktopClient": "Trace"
      },
      "File": {
      "Path": "%LOCALAPPDATA%\\Atlantis\\DC_.log",
      "Append": true,
      "MinLevel": "Trace",
      "FileSizeLimitBytes": 200000000,
      "MaxRollingFiles": 10
      }
   },
   "PbxParameters": "frontsrv,-18321,20000",
   "InstanceName": "#MACHINENAME#",
   "DCWindowMode": "docked",
   "ScalingOptions": "0.6|0.75|0.8|1|1.25|1.5",
   "StartMinimized": false,
   "WebRtc": null,
   "Swp": {
      "SipLogLevel": 0,
      "UseNetworkGuard": true,
      "KeepAliveSipPacketInterval": 50,
      "KeepAliveRtpPacketInterval": 0,
      "PeriodicReconnect": false,
      "PeriodicReconnectInterval": 3600,
      "ConnectionChecker": false,
      "ConnectionCheckerInterval": 55,
      "PortConfiguration": {
      "MaximumRtcpAudioPort": 10088,
      "MinimumRtcpAudioPort": 10000,
      "MinimumRtpAudioPort": 12200,
      "MaximumRtpAudioPort": 12288
      },
      "RegistrationExpiration": 3600,
      "DisplayName": "",
      "AuthName": "",
      "SipServerPort": 5060,
      "LocalSipPort": null,
      "SipPhone_Agent": "Aastra 6867i/5.1.0.2040",
      "JoinConference": false,
      "NeedRegister": true,
      "EnableReliableProvisional": false,
      "WindowScale": 1,
      "ToneEmulation": {
      "BusyHere486": false,
      "BusyHere486Timeout": "00:00:08"
      },
      "Other": {
      "AEC": true,
      "VAD": false,
      "CNG": false,
      "AGC": false,
      "ANS": false,
      "VideoNackStatus": true,
      "DoNotDisturb": false
      },
      "AudioCodecs": {
      "AUDIOCODEC_PCMU": true,
      "AUDIOCODEC_PCMA": true,
      "AUDIOCODEC_G729": true,
      "AUDIOCODEC_ILBC": false,
      "AUDIOCODEC_GSM": false,
      "AUDIOCODEC_AMR": false,
      "AUDIOCODEC_G722": false,
      "AUDIOCODEC_SPEEX": false,
      "AUDIOCODEC_AMRWB": false,
      "AUDIOCODEC_SPEEXWB": false,
      "AUDIOCODEC_G7221": false,
      "AUDIOCODEC_OPUS": false
      },
      "UseMxoneStrategy": false,
      "Workaround": {
      "Use571MxoneExternalCallRefresh": false,
      "UseA5000Workarounds": false,
      "A5000XAastraId": "type\u003d\"23\" model\u003d\"15\" extension\u003d\"000\" version\u003d\"4.2.0.2049\" mac\u003d\"{seededMac}\" GlobalInd\u003d\"15\" SpecificInd\u003d\"\" TxfStatus\u003d\"FFFF\"",
      "ManualAastraMacAddress": "",
      "SeedAstraMacFromExtension": true,
      "UseFakeRegistrationToCheckPassword": false,
      "RegistrationSecretEnabled": true,
      "RegistrationSecret": "987654321",
      "RegistrationSecretWaitTimeMilliseconds": 6000
      },
      "LogDirectory": "%localappdata%\\Atlantis",
      "TransportType": 0,
      "Recording": {},
   },
   "PathCompanyName": "Atlantis"
}
}

DesktopClient Configuration¶

Parameters for configuring application features and connections. In terms of the configuration file, they can be divided into “standalone” and “nested,” where the latter are relevant only when used in a specific section.

Note

The application works either with Swp (SIP softphone) or WebRTC (WebRTC softphone). Therefore, depending on the type of technology used, one section will be inactive (set to null), and the other will be expanded and filled with parameters.

Standalone:

Parameter	Description
PbxParameters	ProServer connection string. values divided by commas, blank spaces among the values, the order is: `<host>,<port>,<timeout>`. mandatory type: string example: `frontsrv,18321,20000` Where: host can be IP or DNS for ProServer 3.15+ the port value should be a negative number (default ProServer port is 18321) timeout in miliseconds
InstanceName	The name of the instance to connect to the ProServer. This is usually the hostname of the remote server. If the DesktopClient and ProServer are on the same machine, you can use the value #MACHINENAME# to replace the hostname of the machine you are running the DesktopClient from (not the remote one).
DebugIndicators	Logging level and it’s settings. Possible values: `App`, `Detail`. Separate the inserted values by commas.
StartMinimized	Settings, which determines the application rendering mode after start. Possible values: `True`, `False` `True` - app started as minimized with minimum possible UI `False` - standard behaviour, ribbon displayed
ScalingOptions	The value list, which will be available as the scaling options for DesktopClient’s ribbon. This can be increased/decreased in proportion to the operating system display settings. Default list: `0.6\|0.75\|0.8\|1\|1.25\|1.5` Inserted numbers will be presented percentually to the user, which means 0.6 = 60%
DCWindowMode	Application bar anchor. `docked` - the bar is docked to the top or bottom of the screen (depending on the user’s settings) `floating` - a floating Windows window that can be moved anywhere
RunOnboardingRoutine	Controls “onboarding” in the application. If `true` the application will attempt to retrieve the configuration from the licensing site, which will then provide the necessary login information. If `false` is deactivated.
OnboardingSuccess	Parameter filled automatically after successful completion of “onboarding”. Values `true` or `false`.
ReactClientOnboardingUrl	Address of the server where the request for login credentials is sent. E.g. `https://fronstage.cz/api/onboarding/accept/dc/onboardingKey`
ReactClientFileResourceUrl	Link where the logo displayed on the splashscreen is downloaded. E.g. `https://frontstage.cz/api/content/logo/fname`
ReactClientTextResourceUrl	Link from where the text displayed on the splashscreen is downloaded. E.g. `https://frontstage.cz/api/content/text/fname`
ProductVersion	Backup value, if the build number is not readable. If not needed, whole tag can be deleted.
StunServerPort	STUN server port. Can be turned off by omitting the value.
PlaybackDeviceVolume	Default volume level for playback device. The higher priority has the user settings tied with the Windows profile. Valid range is 0 - 255.
RecordingDeviceVolume	Default volume level for recording device. The higher priority has the user settings tied with the Windows profile. Valid range is 0 - 255.
UseDirectMakeCallAndAnswer	The setting solves the situation when two active terminals are connected to one workplace, e.g. WebRTC implemented in DesktopClient (hereinafter referred to as DC) and SIP using a mobile client. In this case, terminal conflicts can occur, and the ringing/call sound can be heard on the wrong terminal. `true` - On an outgoing call from the DC, an internal procedure is triggered to pass priority to this terminal (the “StartCall” operation will be routed through the WebRTC operations “makeCall” and “Answer”). `false` - Default behavior where conflicts may occur. The behavior will be the same even if the configuration file does not define the parameter. Important The solution is suitable only where DC is deployed (phone operation via ReactClient or TeamsClient will not work correctly).
UpgradeRequired_User	Sign which determines, whether it’s required to perform the configuration upgrade.
Wnd_Softphone_Topmost	Sign, whether DesktopClient should be always above Windows apps in the operating system.
Wnd_Softphone_ShowInTaskbar	Sign, whether the DesktopCLient app should have the tray icon visible. If False, the icon will be there anyway, but only in the hidden icons area.
ApplicationScale	Scale value for the initial run. If the user then changes the scale, the new value will be saved to `Perso` database table and this parameter is no longer considered. The default is `1`, which means 100% (only one inserted value possible). Related to `ScalingOptions`.
LastUsedCultureCode	Last used localization. If not set the application will set it automatically based on your language. The parameter can be used to force specific localization, for example - `en-US`, `cs-CZ`, and so on.

Swp Section¶

Section dedicated to Built-in softphone and its settings.

Parameter	Description
UserName0	SIP server login.
Password0	SIP server login password.
SipServer	SIP server DNS/IP address.
SipPhone_Agent	`User-Agent` header inserted into SIP messages. Recomended value for: MXONE is `Aastra 6867i/5.1.0.2040` A5000 is `Mediant VE-H SBC/v.7.20A.256.024`
RegistrationExpiration	Registration renewal interval [sec]. Maximum value 3600. Will be inserted into SIP REGISTER message.
KeepAliveSipPacketInterval	SIP keepalive packet sending interval in seconds. This parameter will help in troubleshooting problems with closing connections when inactive. 0 for ban >1 interval in seconds minimum 30 seconds
KeepAliveRtpPacketInterval	RTP keepalive packet sending interval [sec]. It can help troubleshooting the RTP stream closing problem occuring during inactivity. 0 for disabling >1 interval in seconds 15 seconds is minimum
PeriodicReconnect	Whether to register again, after the interval has passed. Ignored, if already registered. Use this parameter only when experiencing network issues and softphone is beign disconnected periodically. values `true` or `false`
PeriodicReconnectInterval	Interval, after the device should be registered again [sec], if the `PeriodicReconnect` is enabled.
ConnectionChecker	Whether to reconnect, after the interval has passed. Softphone will check the connection during a network error (internet or LAN connection). This param can cause trouble when using the A5000 PBX. values `true` or `false`
ConnectionCheckerInterval	Interval [sec], after which reconnect,if the `ConnectionChecker` param. is enabled.
EnableReliableProvisional	Whether to use PRACK. values `true` or `false`
WindowScale	Softphone window scale option. For example 0.8 pro low resolution screens, 1.3 for high resolution screens (in case UI is too small). Recommended value is 1. Tip If you set the value to 0, softphone will be completely hidden. Notification window, during the incomming call will remain vissible.
UseMxoneStrategy	Povinný parameter při použítí MX ONE. Při potížích se zvukem na některých verzích MX ONE, povolte. Jinak ponechte vypnuté, aby se negenerovala zbytečná (nikoli chybná) SIP komunikace.
SipServerPort	SIP port for communication with the server. for softphone with address 10.99.218.81 - port UDP 5060 for softphone with address 10.99.218.84 - port UDP 10000 - 20000
AuthName	Uživatelské jméno pro autorizaci (obvykle stejné jako uživatelské jméno). Některé ústředny něvyžadují nastavení této hodnoty.
DisplayName	User nickname, can be any value. Optional.
SipLogLevel	Log level. `-1` logging not active `1` to `4` determines the level, where `4` means we log everything. If the parameter `LogDirectory` is not set, logging is off regardless of this option.
LogDirectory	Složka pro logovací soubor. Pokud není nastaveno, dojde k vypnutí logování. Cesta by měla existovat a DesktopClient by k ní měl mít přístup. V cestě můžete používat proměnné prostředí, např. `%localappdata%\atlantis\DC_Logs`.
TransportType	Transfer type for SIP communication. Valid values: UDP (for softphones with the addess 10.99.218.81 a 10.99.218.84) TCP TLS At the same time, it is needed that port 18321 is enabled, for the server communiction purposes.
Softphone_PlaybackDeviceIndex	Playback device index
Softphone_RecordingDeviceIndex	Recording device index
Softphone_RecordingDeviceVolume	Recording device volume
RingtoneDeviceId	Ringtone device ID. Can be also set within the DC app - Ribbon designer ‣ Global settings

Port Configuration¶

Port range for RTP and RTC protocol communication. The default is 7002 - 7810 (range 10402-10800 reserved for future use).

Parameter	Description
MaximumRtcpAudioPort	Upper limit of the port range for RTC.
MinimumRtcpAudioPort	Lower limit of the port range for RTC.
MinimumRtpAudioPort	Lower limit of the port range for RTP.
MaximumRtpAudioPort	Upper limit of the port range for RTP.

AudioCodecs¶

Settings for supported voice communication codecs. Activate a codec by setting the parameter to true, and deactivate it by setting it to false.

Parameter	Description
AUDIOCODEC_PCMU	PCMU/G711-law 8KHZ 64kbit/s
AUDIOCODEC_PCMA	PCMA/G711 A-law 8KHZ 64kbit/s
AUDIOCODEC_G729	G729 8KHZ 8kbit/s
AUDIOCODEC_ILBC	iLBC 8KHZ 30ms-13kbit/s 20 ms-15kbit/s
AUDIOCODEC_GSM	GSM 8KHZ 13kbit/s
AUDIOCODEC_AMR	Adaptive Multi-Rate (AMR) 8KHZ (4.75,5.15,5.90,6.70,7.40,7.95,10.20,12.20)kbit/s
AUDIOCODEC_G722	G722 16KHZ 64kbit/s
AUDIOCODEC_SPEEX	SPEEX 8KHZ (2-24)kbit/s
AUDIOCODEC_AMRWB	Adaptive Multi-Rate Wideband (AMR-WB)16KHZ (6.60,8.85,12.65,14.25,15.85,18.25,19.85,23.05,23.85)kbit/s
AUDIOCODEC_SPEEXWB	SPEEX 16KHZ (4-42)kbit/s
AUDIOCODEC_G7221	G722.1 16KHZ (16,24,32)kbit/s
AUDIOCODEC_OPUS	OPUS 48KHZ 32kbit/s

ToneEmulation¶

Parameter	Description
BusyHere486	If the customer is trying to reach an occupied phone, he will hear a “busy tone”. Values `true` or `false`
BusyHere486Timeout	The duration for which the “busy tone” will play. The expected format is HH:MM:SS.

Workaround¶

Parameter	Description
UseA5000Workarounds	Used for troubleshooting when A5000 PBX is used. If you experience “autoanswer/Call.Hold / Call.Unhold” or sound problems, enable this.
Use571MxoneExternalCallRefresh	Obsolete, replaced by `UseMxoneStrategy` When experiencing sound problems on some MX ONE versions, enabling can solve the problem. Otherwise leave off, so no extra SIP communication is being created.
ManualAastraMacAddress	If defined, this MAC address will be used for softphone. Otherwise, local network adapter address (NIC) is used.
A5000AastraId	“X-Aastra-Id” header within SIP communiction. If any is set, it will be used instead of the generated one.
UseFakeRegistrationToCheckPassword	Obsolete. Only for testing purposes, otherwise not recomended to use.
RegistrationSecretEnabled	If allowed, A5000 PBX will send the OK status even when the credentials are not valid. As a part of this settings, configuration items are not sent. This functionality controls the hidden string in configuration item (key defined in `RegistrationSecret`). If not allowed, softphone can generate “false-positive” registration result, even if the action failed.
RegistrationSecret	Key for `RegistrationSecretEnabled`. Must be sent from PBX, if the registration was succesfull. The app checks “NOTIFY” messages with `application/xml` content.
SeedAastraMacFromExtension
RegistrationSecretWaitTimeMilliseconds	When sending the credentials, application waits for here defined interval, if it receives `RegistrationSecret` at least in one A5000 request. Valid range is 500 - 10000 (recomended value is 6000).
UseVirtualizedAudioDevices	For testing purpose on the devices, where physical playback device is not possible (softphone expects any playback device, if not found, it will return a warning message). If on, the function will force the usage of a virtual playback device. After finishing the test, turn this functionality off!

Other¶

Parameter	Description
AEC	Zapnout/Vypnout funkci “Acoustic Echo Cancellation” (Potlačení ozvěny)
VAD	Enable/Disable “Voice Activity Detection” function.
CNG	Enable/Disable “Comfort Noise Generator” function.
AGC	Enable/Disable “Automatic Gain Control” function.
ANS	Enable/Disable “Audio Noise Suppression” function.
VideoNackStatus	Enable/Disable “NACK” function, which hjelps with video quality. Currently not usable.
DoNotDistrurb	Enable/Disable “Do not disturb” mode.

WebRtc Section¶

Configuration for the WebRTC softphone.

Parameter	Description
PhoneType	When AS7 used, the values is always “AS7anCTIupd_DC”
UserName	AS7 user ID with `ctiDeviceId` right assigned
Password	User password, see above
ServerUrl	Direct URL address to the `ancti.js` library which runs the WebRTC backend. AS7 is a multitenant app, so the value for this parameter can be the same for more FS instances.
DeviceName	Terminal and workplace identification. Syntax - `sip:extension_number@term.id_of_terminal`, e.g. `sip:165@term.262`
HoldOnDirectCti	Indication, whether the call HOLD should be performed locally instead of by CSTA operation (temporary solution till release 5.9)
RecordingDeviceId	Windows registered device ID for recording the sound. Optional parameter. If empty, the application will set it automatically after the user selects one.
PlaybackDeviceId	Windows registered device ID for playing the sound. Optional parameter. If empty, the application will set it automatically after the user selects one.
PlaybackDeviceVolume	Volume level of `PlaybackDeviceId`, ranges from 0 to 255 or `NULL`.
RecordingDeviceVolume	Volume level of `RecordingDeviceId`, ranges from 0 to 255 or `NULL`.

OnPremiseStrategy Section¶

This setting affects the sending of onboarding emails for the DesktopClient application as part of Onboarding. If the parameter is NULL (i.e., not filled), a cloud installation of Frontstage is assumed, where Deployment Server is used for distribution. For a local system installation, a filled json structure defining the connection to the SMTP server and the system service is expected.

Note

You can define the settings either in the application’s configuration file or using the configuration parameter OnboardingStrategyJson.

Parameter	Description
DiscoveryHost	Address of the ReactClient service/website including the port it runs on. For example, `https://localhost:5001/`

Smtp¶

SMTP server to be used for email distribution.

Parameter	Description
ServerName	Address of the SMTP server that will process recruitment emails in the case of an “on-premise” installation. The “ServiceAsync” system service must have secure access to SMTP. Otherwise, sending will not work.
ServerPort	SMTP protocol port used by the specified server.
UserName	Login username for the account registered on SMTP.
Password	Password for the account.
From	Sender of the recruitment email as displayed to the recipient.
SslMode	See Rebex documentation.

Logging section¶

DesktopClient can log its activity, which is useful for troubleshooting.

Note

The built-in softphone has own logging.

Important

Enabled logging generates files not bigger than tens of megabytes a day, which consumes some of the computer storage over time. If the computer has a limited storage capacity, it’s necessary to ensure they are periodically deleted. Enabled logging has minimal performance impact on the computer.

Headset call control¶

You can control calls distributed through DesktopClient using the supported headset, where you can use the default button to interact with calls.

Supported:

Jabra Engage 65
Jabra Engage 75

There is no need to install or configure anything, the headset should work from factory settings.

Supported on Windows 10 and Windows 11.

Function of the button in different situations:

Outbound call
- When the call is dialed, the headset switches to the “I’m on a call” state (at that point, the “hang up” operation already works).
- After the customer receives the call, the call can be hung up over the headset. The “mute” function can be used.
Inbound call
- When a call comes in, the headset starts ringing and switches to the “I’m on a call” state.
- You cannot refuse a call; you can only accept it now. After accepting the call, you can press to hang up. “Mute” functionality can be used at this moment.

Troubleshooting¶

The DesktopClient application must be running, and the headset must be connected to your computer for the functionality to work correctly.

If the control still doesn’t work, you can try:

Reset the headset to factory settings. You can find how to do it on each manufacturer’s official website.
For Jabra, you can download the “Jabra Direct” app to check your current settings. The following features must be enabled: