Create a remote phone

AI-generated summary

Genesys Cloud enables creation of remote phone profiles through the Phone Management interface, allowing users to configure remote phones for call handling with specific technical requirements and customizable settings. Users must establish a remote phone base settings profile and possess Telephony > Plugin > All permissions before proceeding. Configuration requires assigning the remote phone number to a Genesys Cloud user profile to ensure call connectivity, as some cell carriers reject calls to unassigned numbers. The Phone tab contains configuration settings that can be customized independently from inherited base settings, with options to enter a phone name, select base settings and site, and specify the remote address using SIP address or E.164 format. Remote phones must maintain an established persistent connection before answering calls, with configurable timeout parameters that specify the duration in seconds before automatic disconnection of idle connections. This feature improves efficiency for subsequent calls by reusing active connections, though remote phones do not support auto-answer functionality. For group ring scenarios, Genesys Cloud employs call analysis to verify delivery, requiring users to provide verbal responses to prevent disconnection. Customization options include Incoming Caller Address settings, which can apply trunk prioritized caller selection rules or pass original caller information as ANI. Call recording settings allow users to enable recording, apply automatic volume leveling to both conversation sides, and select from multiple audio codecs (G726-32, GSM, L16, Opus, PCMA, PCMU, and Truespeech). Dual channel recording is available for specific audio formats (PCMU, PCMA, L16, Opus) and saves each conversation participant's audio to separate streams using designated audio channels. Quality of Service (QoS) configuration is managed through DSCP values placed in the TOS field of RTP packet headers, with a valid range of 00 through 3F. The RTP audio port start range is user-definable between 1024–65,535 (default 4000) for specifying the remote UDP port for recorded packet transmission. Media codec preferences can be prioritized through a preferred codec list in MIME format. DTMF transmission supports two methods: RTP Events (out-of-band processing per RFC 4733, with configurable payload type values 96–127, default 101) or In-band Audio (processing through the audio codec stream). Custom property settings should only be configured as directed by Product Support and require three key parameters: Property Name, Data Type, and Value. Security considerations include avoiding configuration of private phone numbers as remote phones if calls may contain sensitive information subject to regulations such as HIPAA. After configuring all settings, users must click Save Phone to complete and apply the remote phone configuration.

PrerequisitesClick to expand

Once you created a remote phone base settings profile, you can create the phone profile. This profile contains a few settings found on the Phone tab that define the remote phone in Genesys Cloud. Some settings on the Phone tab are inherited from the base settings. However, you can customize the phone by altering any of the settings that it inherited from the base settings configuration-without effecting the original base settings configuration. See Inherited settings.

Notes:

A call to a cell phone configured as a remote phone in Genesys Cloud may not connect if the number is not assigned to a Genesys Cloud user profile. Some cell carriers do not allow calls to unassigned numbers.
If the remote phone is part of a group ring, keep in mind that Genesys Cloud uses call analysis to verify delivery of calls to remote phones. Therefore, to avoid disconnection, users must answer with a verbal response.
Before you can use Genesys Cloud to answer a call on a remote phone, the remote phone must have an established persistent connection.

Warning: Do not set up a remote phone with a private phone number if calls to that phone could include sensitive information, such as health records or credit card data. Doing so could violate certain regulations, such as HIPAA.

Configure the phone

Click Menu > Digital and Telephony > Telephony > Phone Management.
Click the Phones tab.
Click Add.

Phone tab

Configure settings on the Phone tab.

Type a name in the Phone Name box.
From the Base Settings list, select the remote phone base settings configuration you created.
From the Site list, select your site.
In the Remote Address box, enter the phone’s SIP address or remote phone number using E.164 format.
- For more information, see Phone addresses or E.164 overview.
Choose one of the following steps:
- To use the default phone configuration settings and complete the remote phone configuration, click Save Phone.
- To customize the phone configuration settings, proceed to the Customize the Phone section of this article.

Customize the phone

In the Phone Configuration panel, click the arrow to expand the section containing the settings you want to customize.

GeneralClick to expand

Setting	Description
Incoming Caller Address	When you select the Use the trunk’s prioritized caller selection rules option, the Edge uses the default caller ID setting specified by the Prioritized caller ID locations setting in the trunk configuration. When you select the Use the original caller address option, the Edge passed the original caller’s info as the ANI when calling the remote station leg. For more information, see External trunk settings.[/bs_well]
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 handled more efficiently because they can immediately use the existing connection. Note: While remote phones support the persistent connection feature, they do not support the auto answer feature.
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 Genesys Cloud automatically closes it.

MediaClick to expand

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.
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.

CustomClick to expand

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 phone configuration settings and complete the remote phone configuration, click Save Phone.

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Phone tab

Configure settings on the Phone tab.

Type a name in the Phone Name box.

From the Base Settings list, select the remote phone base settings configuration you created.

From the Site list, select your site.

In the Remote Address box, enter the phone’s SIP address or remote phone number using E.164 format.

For more information, see Phone addresses or E.164 overview.

Choose one of the following steps:

To use the default phone configuration settings and complete the remote phone configuration, click Save Phone.
To customize the phone configuration settings, proceed to the Customize the Phone section of this article.

Customize the phone

In the Phone Configuration panel, click the arrow to expand the section containing the settings you want to customize.

GeneralClick to expand

Setting	Description
Incoming Caller Address	When you select the Use the trunk’s prioritized caller selection rules option, the Edge uses the default caller ID setting specified by the Prioritized caller ID locations setting in the trunk configuration. When you select the Use the original caller address option, the Edge passed the original caller’s info as the ANI when calling the remote station leg. For more information, see External trunk settings.[/bs_well]
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 handled more efficiently because they can immediately use the existing connection. Note: While remote phones support the persistent connection feature, they do not support the auto answer feature.
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 Genesys Cloud automatically closes it.

MediaClick to expand

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.
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.

CustomClick to expand

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 phone configuration settings and complete the remote phone configuration, click Save Phone.

Create a remote phone

Configure the phone

Phone tab

Customize the phone

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Create a remote phone

Configure the phone

Phone tab

Customize the phone

[NEXT] Was this article helpful?