Create base settings for a PureCloud softphone
- 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.
Create the base settings
- Click Admin.
- Under Telephony, click 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 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 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). |
| 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. |
- To use your custom base line appearance settings, click Save Base Settings.
- Once you have created a base line settings, you can now Create a PureCloud softphone.
