This reference describes all the settings associated with the AudioCodes 430HD phone.

For information on creating the actual phone configuration in Genesys Cloud, see Create the base settings and Create a phone.

Base Settings

Before you can create a phone, you must create a Base Settings configuration for that phone model. The Base Settings configuration contains a group of settings that define how a particular phone model is to operate in Genesys Cloud. Once you create a Base Settings configuration, you can assign it to a phone with the default settings, or you can customize the settings. This section describes all the settings that you can configure when you choose to customize a Base Settings configuration.

Menu: Telephony / Phone Management / Base Settings / Create-Edit Base Settings

Tab: Base Phone

Section: Phone Configuration

Setting Description
Dynamic Reload

Enabled (Default): Allow the dynamic reloading of the phone configuration.

Disabled: Do not allow dynamic reloading.

Web/TUI Authentication

 Set up an administrative password for configuring the phone from a web-based interface or from the Telephone User Interface/LCD on the phone. By default the password is masked, but you can select the Show Password check box to see the password in plain-text.
Time Settings  
Timezone Discovery

Enabled (Default): Automatically sets the timezone for the phone based on the timezone of the DHCP server.

Disabled : Allows manually setting the timezone based on GMT.
SNTP Server Sets the name of the Simple Network Time Protocol (SNTP) server from which the phone obtains the current time.
GMT Offset

When Timezone Discovery is Enabled, this setting is automatically configured from DHCP.

When Timezone Discovery is Disabled, use this field to set the number of hours to offset the time of the phone from GMT.

Note: When Timezone Discovery is Disabled and Daylight Saving Time is enabled, you should calculate the GMT offset based on the standard timezone.
Daylight Saving Time

Enabled (Default): Allows Daylight Savings Time (DST) to display on the phone. Use the Start and End fields (Week, Day, Month, and Time) to specify the period that DST is in effect. 

Disabled: Do not display DST on the phone.
Setting Description
DSCP Use the list to choose the Differentiated Services Code Point (DSCP) value for Quality of Service (QoS). 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). Default is 2E (46, 101110) EF.
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 16384.
Preferred Codec List Use the list to select and build a list of preferred media codecs in mime format. To change the order in which the codecs in the list are used, use the arrows adjacent to the codec name.
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.
Gain Settings

Use these fields to specify volume level settings, in decibals dB, for the various phone interfaces: Handset, Handsfree, Headset

  • Input Gain: Sets the level of sound picked up by the microphone/mouthpiece.
  • Output Gain: Sets the level of sound delivered to the earpiece.
  • Sidetone Gain: Sets the level of the sound that is picked up by the phone's mouthpiece and instantly transmitted to the earpiece.

Note: It is not advisable to change the default values, unless you are having specific issues. Before you make adjustments, make note of the default vaules.
Input Gain Output Gain Sidetone Gain
Handset The valid range is (-32) to 31 (dB), where -32 is mute. The valid range is (-32) to 31 (dB), where -32 is mute. The valid range is (-32) to 31 (dB), where -32 is mute.
Handsfree The valid range is (-32) to 31 (dB), where -32 is mute. The valid range is (-32) to 31 (dB), where -32 is mute. N/A
Headset The valid range is (-32) to 31 (dB), where -32 is mute. The valid range is (-32) to 31 (dB), where -32 is mute. The valid range is (-32) to 31 (dB), where -32 is mute.

Setting Description
Provisioning
Provision Source 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.
Custom Configuration Files

Allows specifying a location for a custom configuration file containing additional provisioning information for phones. The information in a custom configuration file is appended to the provisioning information already passed to the phone. The content and format of a custom configuration file is the administrator's responsibility. The Custom Configuration Files field just allows you to point to the file's location.

For more information, see the appropriate Administrator's Manual for your phone in the AudioCodes Library.

Note: Attributes defined in the custom configuration file cannot be used to override attributes previously defined by the standard configuration file.

Setting Description
Message Waiting Indicator

Use this setting to configure the Message Waiting Indicator (MWI), the light that blinks to indicate that the user has a new voicemail.

To enable the feature, select the Message Waiting Indicator (MWI) check box.

For more information, see Configure the Message Waiting Indicator setting.
Inter-digit Delay

The duration after each digit is entered before the system assumes that the user has finished entering digits.
Setting Description
System Logging (Syslog)

Use the toggle to enable or disable a phone's ability to log data.

Disabled (Default): Do not log data.

Enabled: Log data to either an Edge or a Syslog Server. Once enabled, the following fields need to be filled in.

Note: When you enable this setting, you must also enable Phone System Logging in the Diagnostic>Phone System Logging section of the Phone Trunk Connection Configuration. See Create a SIP phone connection trunk.

Send Syslogs to Edges

Configure the phone to send data to an Edge.

Note: When you choose this option, the Send Syslogs to Syslog Server setting is disabled. However, you must set the Syslog Port Number to either the default port number (514) or to the port number specified in the Diagnostic>Phone System Logging section of the Phone Trunk Connection Configuration.
Send Syslogs to Syslog Server

Configure the phone to send data to a Syslog Server.

Note: When you choose this option, you must specify both the Syslog Server Address and the Syslog Port Number.
Syslog Server Address Specify the address of the server to which you want to send the logs.
Syslog Server Port  Specify the port number on the server that is configured to receive the logs.
Trace Levels

Severity level settings for each option are as follows:

  • None: Do not log.
  • Emergency: Indicates that the system is unusable.
  • Error: Indicates that error conditions exist.
  • Warning: Indicates that an error might occur if measures are not taken to prevent it.
  • Notice: Indicates that an unusual event has occurred.
  • Info: Indicates an operational message.
  • Debug: Provides Information useful to developers for debugging purposes.
802.1x Defines the level at which Syslog messages are generated related to the security protocol.
Control Center Defines the level at which Syslog messages are generated related to Networking.
DSP Defines the level at which Syslog messages are generated related to the voice engine of the phone.
Kernel Defines the level at which Syslog messages are generated related to the kernel process of the phone.
LCD Display Defines the level at which Syslog messages are generated related to LCD Display and other keypresses.
SIP Call Control Defines the level at which Syslog messages are generated related to MTR layer Radvision.
SIP Stack Defines the level at which Syslog messages are generated related to SIP Stack Radvision.
VoIP Application Defines the level at which Syslog messages are generated related to the VoIP application.
Watchdog Defines the level at which Syslog messages are generated related to Watchdog, which is responsible for keeping other processes running.
Web Defines the level at which Syslog messages are generated related to phone web server.
Network Packet Recording Select the check box adjacent to the option to enable recording of the associated item.
TDM In Recording Activates the DSP TDM (TDM In) recording of incoming audio traffic.
TDM Out Recording Activates the DSP network (TDM Out) recording of outgoing audio traffic.
Echo Canceller Debug Recording Activates the recording of the echo cancellation activity to debug associated echo problems. 
RTP Recording Activates the DSP RTP recording.
Packet Recording Activates the packet recording mechanism.
Server Address The IP address of the remote computer to which the recorded packets are sent.
Port Number Defines the UDP port of the remote computer to which the recorded packets are sent.
The Custom option is designed to allow Genesys Cloud Customer Care personnel to alter a phone configuration for troubleshooting or special circumstances. 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.

Tab: Base Line Appearance

Section: Configuration

Setting Description
Calls Per Line The number of calls that this line can handle.
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.
Proxy Keep Alive Timer Defines the proxy keep alive time interval (in seconds) between keep alive messages.
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 adjacent 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.
Setting Description
Auto-Dial Settings
Enable Auto-Dial

Off (Default): Do not use auto-dial settings.

On: Use the following fields to specify the auto-dial settings.
Auto-Dial Destination The address to dial. This could be a SIP URI or telephone number.
Auto-Dial timeout The delay (in seconds) after going off-hook before dialing the destination. An example is a lobby phone that auto-dials the receptionist. Valid range is 0-120 seconds for automatically dialing out the number entered.
Auto-Conference Settings When this setting is enabled, and if a call is already connected or held at the station, a conference is created between the new incoming call and the existing call(s). An announcement of the new call is played to the existing call(s) before the conference is established.
Enable Auto-Conference

Off (Default): Do not use auto-conference settings.

On: Use the following fields to specify the auto-conference settings.
Auto-Conference PIN Required number to join a conference.
Language Speech language selected to use for the conference voice menu.

Phone Settings

When you create a phone and assign a base settings configuration to it, the phone is essentially ready for use in Genesys Cloud. However, if you want, you can override the inherited base settings and customize the settings for a specific phone. This section describes all the settings that you can configure when you choose to customize a specific phone. 

Menu: Telephony / Phone Management / Phones / Create-Edit Phone

Tab: Phone

Section: Phone Configuration

Setting Description
Dynamic Reload

Enabled (Default): allows the dynamic reloading of the phone configuration.

Disabled: Do not allow dynamic reloading.

Web/TUI Authentication

 Set up an administrative password for configuring the phone from a web-based interface or from the Telephone User Interface/LCD on the phone. By default the password is masked, but you can select the Show Password check box to see the password in plain-text.
Time Settings  
Timezone Discovery

Enabled (Default): Automatically sets the timezone for the phone based on the timezone of the DHCP server.

Disabled : Allows manually setting the timezone based on GMT.
SNTP Server Sets the name of the Simple Network Time Protocol (SNTP) server from which the phone obtains the current time.
GMT Offset

When Timezone Discovery is Enabled, this setting is automatically configured from DHCP.

When Timezone Discovery is Disabled, use this field to set the number of hours to offset the time of the phone from GMT.

Note: When Timezone Discovery is Disabled and Daylight Saving Time is enabled, you should calculate the GMT offset based on the standard timezone.
Daylight Saving Time

Enabled (Default): Allows Daylight Savings Time (DST) to display on the phone. Use the Start and End fields (Week, Day, Month, and Time) to specify the period that DST is in effect. 

Disabled: Do not display DST on the phone.
Setting Description
DSCP Use the list to choose the Differentiated Services Code Point (DSCP) value for Quality of Service (QoS). 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). Default is 2E (46, 101110) EF.
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 16384.
Preferred Codec List Use the list to select and build a list of preferred media codecs in mime format. Use the arrows adjacent 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.
Gain Settings

Use these fields to specify volume level settings, in decibals dB, for the various phone interfaces: Handset, Handsfree, Headset

  • Input Gain: Sets the level of sound picked up by the microphone/mouthpiece.
  • Output Gain: Sets the level of sound delivered to the earpiece.
  • Sidetone Gain: Sets the level of the sound that is picked up by the phone's mouthpiece and instantly transmitted to the earpiece.

Note: It is not advisable to change the default values, unless you are having specific issues. Before you make adjustments, make note of the default vaules.
Input Gain Output Gain Sidetone Gain
Handset The valid range is (-32) to 31 (dB), where -32 is mute. The valid range is (-32) to 31 (dB), where -32 is mute. The valid range is (-32) to 31 (dB), where -32 is mute.
Handsfree The valid range is (-32) to 31 (dB), where -32 is mute. The valid range is (-32) to 31 (dB), where -32 is mute. N/A
Headset The valid range is (-32) to 31 (dB), where -32 is mute. The valid range is (-32) to 31 (dB), where -32 is mute. The valid range is (-32) to 31 (dB), where -32 is mute.


Setting Description
Provisioning
Provision Source 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.
Custom Configuration Files

Allows specifying a location for a custom configuration file containing additional provisioning information for phones. The information in a custom configuration file is appended to the provisioning information already passed to the phone. The content and format of a custom configuration file is the administrator's responsibility. The Custom Configuration Files field just allows you to point to the file's location.

For more information, see the appropriate Administrator's Manual for your phone in the AudioCodes Library.

Note: Attributes defined in the custom configuration file cannot be used to override attributes previously defined by the standard configuration file.
Signaling
DSCP Use the list to choose the Differentiated Services Code Point (DSCP) value of Quality of Service (QoS). 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
Message Waiting Indicator

Use this setting to configure the Message Waiting Indicator (MWI), the light that blinks to indicate that the user has a new voicemail.

To enable the feature, select the Message Waiting Indicator (MWI) check box.

For more information, see Configure the Message Waiting Indicator setting.
Inter-digit Delay

The duration after each digit is entered before the system assumes that the user has finished entering digits.
Setting Description
System Logging (Syslog)

Use the toggle to enable or disable a phone's ability to log data.

Disabled (Default): Do not log data.

Enabled: Log data to either an Edge or a Syslog Server. Once enabled, the following fields need to be filled in.

Note: When you enable this setting, you must also enable Phone System Logging in the Diagnostic>Phone System Logging section of the Phone Trunk Connection Configuration. See Create a SIP phone connection trunk.

Send Syslogs to Edges

Configure the phone to send data to an Edge.

Note: When you choose this option, the Send Syslogs to Syslog Server setting is disabled. However, you must set the Syslog Port Number to either the default port number (514) or to the port number specified in the Diagnostic>Phone System Logging section of the Phone Trunk Connection Configuration.
Send Syslogs to Syslog Server

Configure the phone to send data to a Syslog Server.

Note: When you choose this option, you must specify both the Syslog Server Address and the Syslog Port Number.
Syslog Server Address Specify the address of the server where the syslog exists.
Syslog Server Port  Specify the port number to use to access the server where the syslog exists.
Trace Levels

Severity level settings for each option are as follows:

  • None: Do not log.
  • Emergency: Indicates that the system is unusable.
  • Error: Indicates that error conditions exist.
  • Warning: Indicates that an error might occur if measures are not taken to prevent it.
  • Notice: Indicates that an unusual event has occurred.
  • Info: Indicates an operational message.
  • Debug: Provides Information useful to developers for debugging purposes.
802.1x Defines the level at which Syslog messages are generated related to the security protocol.
Control Center Defines the level at which Syslog messages are generated related to Networking.
DSP Defines the level at which Syslog messages are generated related to the voice engine of the phone.
Kernel Defines the level at which Syslog messages are generated related to the kernel process of the phone.
LCD Display Defines the level at which Syslog messages are generated related to LCD Display and other keypresses.
SIP Call Control Defines the level at which Syslog messages are generated related to MTR layer Radvision.
SIP Stack Defines the level at which Syslog messages are generated related to SIP Stack Radvision.
VoIP Application Defines the level at which Syslog messages are generated related to the VoIP application.
Watchdog Defines the level at which Syslog messages are generated related to Watchdog, which is responsible for keeping other processes running.
Web Defines the level at which Syslog messages are generated related to phone web server.
Network Packet Recording Select the check box adjacent to the option to enable recording of the associated item.
TDM In Recording Activates the DSP TDM (TDM In) recording of incoming audio traffic.
TDM Out Recording Activates the DSP network (TDM Out) recording of outgoing audio traffic.
Echo Canceller Debug Recording Activates the recording of the echo cancellation activity to debug associated echo problems.
RTP Recording Activates the DSP RTP recording.
Packet Recording Activates packet recording mechanism.
Server Address The IP address of the remote computer to which the recorded packets are sent.
Port Number Defines the UDP port of the remote computer to which the recorded packets are sent.
The Custom option is designed to allow Genesys Cloud Customer Care personnel to alter a phone configuration for troubleshooting or special circumstances. 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.

Tab: Line Keys

Section: Configuration

Setting Description
Calls Per Line The number of calls that this line can handle.
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.
Proxy Keep Alive Timer Defines the proxy keep alive time interval (in seconds) between keep alive messages.
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 adjacent 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.
Setting Description
Auto-Dial Settings
Enable Auto-Dial

Off (Default): Do not use auto-dial settings.

On: Use the following fields to specify the auto-dial settings.
Auto-Dial Destination The address to dial. This could be a SIP URI or telephone number.
Auto-Dial timeout The delay (in seconds) after going off-hook before dialing the destination. An example is a lobby phone that auto-dials the receptionist. Valid range is 0-120 seconds for automatically dialing out the number entered.
Auto-Conference Settings When this setting is enabled, and if a call is already connected or held at the station, a conference is created between the new incoming call and the existing call(s). An announcement of the new call is played to the existing call(s) before the conference is established.
Enable Auto-Conference

Off (Default): Do not use auto-conference settings.

On: Use the following fields to specify the auto-conference settings.
Auto-Conference PIN Required number to join a conference.
Language Speech language selected to use for the conference voice menu.
The Custom option is designed to allow Genesys Cloud Customer Care personnel to alter a phone configuration for troubleshooting or special circumstances. 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.