Create base settings for the Spectralink wireless

PrerequisitesClick to expand

You can configure the Spectralink wireless phone to work in Genesys Cloud by creating a base settings profile. This profile contains a group of settings found on the Base Phone and Base Line Appearance tabs that define how a Spectralink wireless phone is to operate in Genesys Cloud. Once you create a base settings configuration, you can save it with the default settings or you can customize the settings.

Click Admin.
Under Telephony, click Phone Management.
Click Menu > Channels > Telephony > Phone Management.
Click the Base Settings tab.
Click Add Base Settings.

Base Phone tab

Configure settings on the Base Phone tab.

Type a name in the Base Settings Name box.
From the Phone Make and Model list, select Spectralink wireless.
Leave Standalone Features set to Off unless you are creating a base settings configuration for conference room phones. See Enable standalone features.
Choose one of the following steps:

To use the default base phone settings, click Save Base Settings and proceed to the Configure the base line appearance section of this article.
To customize the base settings, proceed to the Customize the base settings section of this article.

Customize the base settings

In the Phone Configuration section of the Base Phone tab, there are six expandable sections: General, Media, Network, Interface, Diagnostic, and Custom that allow you to change the base settings for the Spectralink wireless phone.

To customize the base settings:

Select the appropriate settings from each expandable section.

GeneralClick to expand

Setting	Description
Dynamic Reload	Enabled (Default): allows the dynamic reloading of the phone configuration. Disabled: Do not allow dynamic reloading.
Persist User Settings	Enabled (Default): Persist those users settings that are configured via a web-based interface or from the Telephone User Interface/LCD on the phone. Disabled: Do not persist user settings.
Web/TUI Authentication	Set up an administrative password for configuring the phone from a web-based interface or from the Telephone User Interface/LCD on the phone. By default the password is masked, but you can select the Show Password check box to see the password in plain-text.
Time Settings
GMT Offset	Use this field to set the number of hours to offset the time of the phone from GMT.
SNTP Server	Sets the name of the Simple Network Time Protocol (SNTP) server from which the phone obtains the current time.
Daylight Saving Time	Enabled (Default): Allows Daylight Savings Time (DST) to display on the phone. Use the Start and End fields (Week, Day, Month, and Time) to specify the period that DST is in effect. Disabled: Do not display DST on the phone.

MediaClick to expand

Setting	Description
DSCP	Use the list to choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) for RTP packets. The system places this value in the upper 6 bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every RTP packet. The range of values available is 00 (0,000000) through 3F (63, 111111).
RTP Audio Port Start Range	Defines the UDP port of the remote computer where the system sends the recorded packets. The valid range is 1024–65,535. The default port is 4000.
Preferred Codec List	Use the list to select and build a list of preferred media codecs in mime format. Use the arrows next to the codec name to change the order in which the codecs in the list are used.
DTMF Settings
DTMF Payload	Specify the payload type value to use when the DTMF Method type is set to RTP Events. Valid range is 96–127. Valid only when DTMF Method value is set to RTP Events. The default value is 101.
DTMF Method	Use the list to select the method to use to transmit dual-tone multifrequency (DTMF) signaling. Select RTP Events to enable out-of-band processing of events from the RTP stream (RFC 4733). Select In-band Audio for the processing, detection, and synthesis from the audio codec stream. The default value is RTP Events.

NetworkClick to expand

Setting	Description
Provisioning	Use the Provision list to select the source that the phone uses to obtain the provision configuration data.
Provision Source	Use the Provision Source list to select the source that the phone uses to obtain the provision configuration data.
From Edges within the Site	Configure the phone to obtain provision configuration data from an Edge on your site.
From a third party URI	Configure the phone to obtain provision configuration data from a third-party URI.
Provisioning Third Party URI	When you select From a third party URI in the Provision list, this box is enabled. Enter the URI to the third-party resource that the phone uses to obtain provision configuration data.
TLS Authority ID	Specify the trusted certificate authority to validate connections when using TLS.
Signaling
DSCP	Use the list to choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) for SIP packets. The system places this value in the upper 6 bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every SIP packet. The range of values available is 00 (0,000000) through 3F (63, 111111).
TCP Keep-Alive Messages	Use this setting to configure the sending of keepalive messages between the Edge/Genesys Cloud and the phone. Keepalive messages are probe packets that contain no data but have the ACK flag turned on; forcing the phone to respond. Disabled (default): Disables the sending of keepalive messages. Enabled: Periodically send the phone keepalive messages. When you enable the TCP Keep-Alive Messages, set the number of seconds in the Every box to specify how often you want the Edge/Genesys Cloud to send keepalive messages. The default is every 30 seconds. The valid range is 10–7,200. Note: Setting a keep-alive timer generates extra network traffic.

InterfaceClick to expand

Setting	Description
Message Waiting Indicator	Use this setting to configure the Message Waiting Indicator (MWI), the light that blinks to indicate that the user has a new voicemail. To enable the feature, select the Message Waiting Indicator (MWI) check box. For more information, see Configure the Message Waiting Indicator setting.
DND Sync	Disabled (default): Disables Do Not Disturb (DND) synchronization between Edge and the phone. Enabled: Enables DND synchronization between Edge and the phone.

Setting

Description

Message Waiting Indicator

Use this setting to configure the Message Waiting Indicator (MWI), the light that blinks to indicate that the user has a new voicemail.

To enable the feature, select the Message Waiting Indicator (MWI) check box.

For more information, see Configure the Message Waiting Indicator setting.

DND Sync

Disabled (default): Disables Do Not Disturb (DND) synchronization between Edge and the phone.

Enabled: Enables DND synchronization between Edge and the phone.

DiagnosticClick to expand

Setting	Description
Trace Levels	Severity level settings for each option are as follows: Fatal error: The tracing shows those errors that immediately cause the system to crash. Major error: The tracing shows those errors that will eventually incapacitate the system. Minor error: The tracing shows only those errors from which the system recovers gracefully. Low detail class event: The tracing includes the lowest amount detail about the class of event that you selected. Moderate detail class event: The tracing includes a medium amount of detail about the class of event that you selected. High detail class event: The tracing includes significant detail about the class of event that you selected. Debug: The highest level of tracing available.
Application	Sets the tracing level for the Application syslog topic.
Configuration	Sets the tracing level for the Configuration syslog topic.
Micro Browser	Sets the tracing level for the Micro Browser syslog topic.
Copy	Sets the tracing level for the Copy syslog topic.
Curl	Sets the tracing level for the CURL syslog topic.
Key	Sets the tracing level for the Key syslog topic.
SIP	Sets the tracing level for the SIP syslog topic.
Support Objects	Sets the tracing level for the Support Objects syslog topic.
TLS	Sets the tracing level for the TLS syslog topic.
Wapp Mgr	Sets the tracing level for the Wapp Mgr syslog topic.

CustomClick to expand

The Custom option is designed to allow Genesys Cloud Product Support personnel to alter a phone configuration for troubleshooting or special circumstances. Only enter custom property settings as directed by Genesys Cloud Product Support.

Setting	Description
Property Name	The name to assign to the custom property.
Data Type	The data type for the custom property.
Value	The value to assign the custom property.

To use your custom base settings, click Save Base Settings and proceed to the Configure the base line appearance section of this article.

Configure the base line appearance

Click the Base Line Appearance tab.
Type a name in the Key Label box.
Choose one of the following steps:
- To use the default base line appearance settings, click Save Base Settings. You can now Create a Spectralink wireless phone.
- To customize the base line appearance settings, proceed to the Customize the base line appearance section of this article.

Customize the base line appearance

In the Configuration setion of the Base Line Appearance tab, three expandable sections appear: General, Signaling, and Interface. Each expandable section contains controls that customize the settings for the PureCloud Softphone.

To customize the base line appearance settings:

Select the appropriate settings from each expandable section.

GeneralClick to expand

Setting	Description
Calls Per Line	The number of calls that this line can handle.
Persistent Connection Settings	When the persistent connection feature is disabled, Genesys Cloud must create a connection for every call. When you enable the persistent connection feature and set a timeout value, you improve Genesys Cloud’s ability to process subsequent calls. More specifically, calls that come in while the connection is still active are immediately alerted via the UI or are auto answered if the Auto Answer feature is configured for the user.
Enable Persistent Connection	Disabled (Default): Do not use persistent connection feature. Enabled: Turn on persistent connection feature.
Persistent Connection Timeout	Sets the amount of time, in seconds, that the open connection can remain idle before Genesys Cloud automatically closes it.

MediaClick to expand

Setting	Description
Recording Settings
Record calls on this phone	Use this check box to enable or disable recording. The default setting is disabled.
Level the volume on both sides of the conversation	Use this check box to enable or disable automatic level control for the recordings. The default setting is disabled.
Audio Format/Codec	Use this list to select the audio codec to use for recording. The available choices are: G726-32 GSM L16 (Uncompressed Linear 16-bit PCM) Opus PCMA (G.711 A-law) PCMU (G.711 u-law) Truespeech
Dual channel	Use this check box to enable or disable dual channel recording. The Dual channel setting is only available if you select one of the following audio formats: PCMU (G.711 u-law) PCMA (G.711 A-law) L16 (Uncompressed Linear 16-bit PCM) Opus When you enable the Dual channel setting, the system saves each channel of the recording in a separate stream. The system uses audio channel 0 to save the recording connected to the phone and audio channel 1 to save the recording connected to the other party. The default setting is enabled.
DSCP	Use the list to choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS) for RTP packets. The system places this value in the upper 6 bits of the TOS (Type Of Service) field. The TOS field is in the IP header of every RTP packet. The range of values available is 00 (0,000000) through 3F (63, 111111).
RTP Audio Port Start Range	Defines the UDP port of the remote computer where the system sends the recorded packets. The valid range is 1024–65,535. The default port is 4000.
Preferred Codec List	Use the list to select and build a list of preferred media codecs in mime format. Use the arrows next to the codec name to change the order in which the codecs in the list are used.
DTMF Settings
DTMF Payload	Specify the payload type value to use when the DTMF Method type is set to RTP Events. Valid range is 96–127. Valid only when DTMF Method value is set to RTP Events. The default value is 101.
DTMF Method	Use the list to select the method to use to transmit dual-tone multifrequency (DTMF) signaling. Select RTP Events to enable out-of-band processing of events from the RTP stream (RFC 4733). Select In-band Audio for the processing, detection, and synthesis from the audio codec stream. The default value is RTP Events.

SignalingClick to expand

Setting	Description
Protocol	Use the list to select the SIP protocol the phone uses to register: UDP, TCP, or TLS. The default is UDP.
Listen Port	Defines the local SIP listen port for SIP messages. The listening ports are the network ports on which the station expects to receive messages from SIP peers. While you can enter a custom port number, each protocol has a default listen port.
Registration period	The periodic delay (in seconds) between sending a SIP REGISTER).
Max Bindings	Specifies the maximum number of bindings.
Proxy Keep Alive Timer	Defines the proxy keep alive time interval (in seconds) between keep alive messages.
SIP Servers or Proxies	The SIP Servers or Proxies setting is not configurable for the first line appearance to ensure that Genesys Cloud sends all outbound requests to the Edge. More specifically, this default configuration for the first line appearance ensures that the phone uses the Edge for the default call control. As long as you did not enable the Span appearance to remaining keys setting, you can choose to define where Genesys Cloud sends outbound requests for the remaining Line Keys. You can choose the Edge or specify custom SIP server or proxy server addresses and ports.
Use Edge	Send all outbound requests to the Edge.
Use the following	Use the Hostname or IP Address and Port fields to construct a prioritized list of SIP servers or proxy servers to use to process outbound requests. Use the + to add the server to the list. Use the arrows next to the address name to change the order in which the servers in the list are used.
Digest Authentication	When outbound requests are challenged with digest authentication, use the following credentials:
User Name	The user name pushed down to the phone that changes the password on the administrative account for the physical phone menus.
Password	The password pushed down to the phone that changes the password on the administrative account for the physical phone menus. By default the password is masked, but you can select the Show Password check box to see the password in plain-text.

InterfaceClick to expand

Setting	Description
Auto-Conference Settings	When this setting is enabled, and if a call is already connected or held at the station, a conference is created between the new incoming call and the existing call. An announcement of the new call is played to the existing call before the conference is established.
Enable Auto-Conference	Off (Default): Do not use auto-conference settings. On: Use the following fields to specify the auto-conference settings.
Auto-Conference PIN	Required number to join a conference.
Language	Speech language selected to use for the conference voice menu.

To use your custom base line appearance settings, click Save Base Settings.
Once you have created base line settings for the Spectralink, you can Create a Spectralink wireless phone.

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