Configure the base line appearance settings for the PureCloud softphone
- Click Admin > Telephony.
- Click Phone Management.
- From the Base Phone tab, select the Base Line Appearance tab.
- In the Key Label field, enter a name.
- To save a base setting with the default settings, select the Save Base Settings button.
- 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 the Save Base Settings button.
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. Disabled (Default): Do not use persistent connection feature. Enabled: Turn on persistent connection feature.
Setting
Description
Calls Per Line
The number of calls that this line can handle.
Persistent Connection Settings
Enable Persistent Connection
Persistent Connection Timeout
Sets the amount of time, in seconds, that the open connection can remain idle before being automatically closed.
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. |
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. |