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.
Create the base settings
Click Admin.
Under Telephony, click Phone Management.
Click Menu > Digital and Telephony > Telephony > Phone Management.
Click the Base Settings tab.
Click Add Base Settings.
Type a name in the Base Settings Name box.
From the Phone Make and Model list, select PureCloud Softphone.
Choose one of the following steps:
To use the default base phone settings, click Save Base Settingsand 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 settings
In the Phone Configuration section of the Base Phone tab, there are four expandable sections: General, Media, Network, and Custom that allow you to change the base settings for the Genesys Cloud WebRTC phone.
To customize the base settings:
Select the appropriate settings from each expandable section.
Setting
Description
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
Provisioning
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).
The Custom option is designed to allow Genesys Cloud Product Support personnel to alter a phone configuration for troubleshooting or special circumstances. You should 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 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
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 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
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.
Setting
Description
Calls Per Line
The number of ACD 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.
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.
To use your custom base line appearance settings, click Save Base Settings.
