Create base settings for the Spectralink wireless
- Telephony > Plugin > All permission
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 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.
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.
|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.|
|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.
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.|
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.
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.
|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.
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).
|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.
The Message Waiting Indicator has three options:
Disabled (default): Disables Do Not Disturb (DND) synchronization between Edge and the phone.
Enabled: Enables DND synchronization between Edge and the phone.
Severity level settings for each option are as follows:
|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.|
|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.
- Chose 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.
|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 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.|
|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.|
|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.|
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 a base line settings for the Spectralink, you can Create a Spectralink wireless phone.