Create base settings for a PureCloud softphone


Prerequisites

  • Telephony > Plugin > All permission

Before you can create a PureCloud softphone, you create a base settings configuration. The base settings configuration contains a group of settings found on the Base Phone and Base Line Appearance tabs that define how a PureCloud softphone 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.

Note: The PureCloud Softphone is different from the Genesys Cloud WebRTC phone. For more information, see About Genesys Cloud WebRTC phones.

Configure the base phone

  1. Click Admin. 
  2. Under Telephony, click Phone Management.
  3. Click the Base Settings tab.
  4. Click Create New and the Base Phone tab appears.PureCloud Softphone Base Settings save
  5. Type a name in the Base Settings Name field.
  6. From the Phone Make and Model list, select PureCloud Softphone.
  7. Perform one of the following:
    • 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 phone settings, proceed to the Customize the base phone section of this article.

Customize the base phone

  1. In the Phone Configuration panel, click the arrow to expand the section containing the settings you want to customize
Setting Description
Firmware Version

The versions of firmware that Genesys Cloud supports for the make and model of that phone. You can use the list to choose from the different versions supported.

Dynamic Reload Enable the dynamic reloading of the phone configuration.

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.

Setting Description
Provision Use the Provision 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).

The Custom option is designed to allow Genesys Cloud Customer Care personnel to alter a phone configuration for troubleshooting or special circumstances. You should only enter custom property settings as directed by Genesys Cloud Customer Care.
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.

  1. To use the 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

  1. Click the Base Line Appearance tab.PureCloud Softphone Base Line Save
  2. Type a name in the Key Label field.
  3. Perform one of the following:
    • To use the default base line appearance settings, click Save Base Settings. You can now Create a PureCloud softphone.
    • To customize the base line appearance settings, proceed to the Customize the base line appearance section of this article.

Customize the base line appearance

  1. In the Configuration panel, click the arrow to expand the section containing the settings you wish to customize.
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.

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.
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.
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.


  1. To use your custom base line appearance settings, click Save Base Settings.
  2. Once you have created a base line settings, you can now Create a PureCloud softphone.