Configure the base line appearance settings for the PureCloud softphone

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Genesys Cloud enables configuring base line appearance settings for the PureCloud softphone, including calls per line, persistent connection settings, signaling protocols, UDP settings, interface options, and auto-conference capabilities. Key capabilities include defining calls per line, enabling persistent connections to improve call processing, configuring SIP protocol and ports, setting SIP server registration and authentication, customizing UDP timeouts and retries, and enabling auto-conference features like PIN numbers and voice menus. The settings allow optimizing and customizing the PureCloud softphone for an organization's needs.

Click Admin.
Under Telephony, click Phone Management.
Click Menu > Digital and Telephony > Telephony > Phone Management.
From the Base Phone tab, select the Base Line Appearance tab.
In the Key Label box, enter a name.
To save a base setting with the default settings, click Save Base Settings.
To save a base setting with custom settings, select the arrow next to the Configuration topic you want to configure.
Modify the settings. For more information, refer to the sections below.
Select Save Base Settings.

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 being automatically closed.

SignalingClick to expand

Setting	Description
Protocol	Use the drop-down 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.
SIP Servers or Proxies	Choose an option to define where all outbound requests are to be sent.
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.
UDP Settings
UDP T1 timeout	The timer value that represents the initial incremental delay between UDP packet retransmission.
UDP T2 timeout	The timer value that represents the maximum incremental delay between packet retransmissions.
Max Packet Retry	Maximum number of times to retransmit a SIP message over UDP.
Max Invite Retry	Maximum number of times to retransmit a SIP INVITE request over UDP.

InterfaceClick to expand

Setting	Description
Auto-Conference Settings
Enable Auto-Conference	Off (Default): Do not use auto-dial settings. On: Use the following fields to specify the auto-dial settings.
Auto-Conference PIN	Number required to join a conference.
Language	Speech selected to use for the conference voice menu.

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Configure the base line appearance settings for the PureCloud softphone