Polycom SoundPoint IP 330 settings
This reference describes all the settings associated with the Polycom SoundPoint IP 330 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 | Disabled (Default): Allows manually setting the timezone difference. Enabled: Automatically sets the timezone for the phone based on the timezone of the DHCP server. |
| SNTP Server | Sets the name of the Simple Network Time Protocol (SNTP) server from which the phone obtains the current time. |
| GMT Offset | Sets the number of hours to offset the time of the phone from GMT. Note: When you use the Daylight Saving Time setting, 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 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 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 next to the codec name. |
| DTMF Settings | |
| DTMF Payload | Specify the payload type value to use when the DTMF Method type is 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 | |
| Provision Source | Use the Provision Source list to select the source that the phone uses to obtain the provision configuration data.
|
| 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 Working with Configuration Files in the Poly Documentation Library. Note: Attributes defined in the custom configuration file cannot be used to override attributes previously defined by the standard configuration file. Warning: When using the Custom Configuration Files feature, it is possible to inadvertently introduce a setting that is incompatible with Genesys Cloud which may cause the phone to malfunction. If this happens, it is the administrator's responsibility to find and remove the offending setting to restore functionality. |
| Custom Directory | Specify a corporate directory file containing contact information for people in your company. For more information, see Corporate Directory page in the Poly Documentation Library. |
| 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). |
| TCP Keep-Alive Messages | Use this setting to configure the sending of keepalive messages between the Edge/Genesys Cloud and the phone. Keepalive messages are probe packets that contain no data but have the ACK flag turned on; forcing the phone to respond. Disabled (default): Disables the sending of keepalive messages. When you enable the TCP Keep-Alive Messages, set the number of seconds in the Every box to specify how often you want the Edge/Genesys Cloud to send keepalive messages. The default is every 30 seconds. The valid range is 10-7,200. Note: Setting a keep-alive timer generates extra network traffic. |
| Strict User Validation | Use this feature to control how the phone handles incoming calls and to strengthen the security of the phone. To enable the feature, select the Phone only accepts SIP INVITE requests from Genesys registration servers option. After this this setting is enabled, the phone will ignore any SIP messages that do not come from the Genesys SIP servers to which the phone is registered. Warning: Enabling this setting could interfere with remote site survivability configurations. More specifically, do not enable this setting if your phones connect to an external device for site survivability. The Do not validate SIP requests to the phone disables this feature and is the default setting. |
| 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. Select the Message Waiting Indicator (MWI) check box to enable the feature. For more information, see Configure the Message Waiting Indicator setting. |
| DND Sync | Disabled (Default): Disables Do Not Disturb (DND) synchronization between Edge and the phone. Enabled: Enables DND synchronization between Edge and the phone. |
| 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 will 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. |
| 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:
|
| Application | Sets the tracing level for the Application syslog topic. |
| Configuration | Sets the tracing level for the Configuration syslog topic. |
| Micro Browser | Sets the tracing level for the Micro Browser syslog topic. |
| Copy | Sets the tracing level for the Copy syslog topic. |
| Curl | Sets the tracing level for the CURL syslog topic. |
| Key | Sets the tracing level for the Key syslog topic. |
| SIP | Sets the tracing level for the SIP syslog topic. |
| Support Objects | Sets the tracing level for the Support Objects syslog topic. |
| TLS | Sets the tracing level for the TLS syslog topic. |
| Wapp Mgr | Sets the tracing level for the Wapp Mgr syslog topic. |
| 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. |
| 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 |
|---|---|
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:
|
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:
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. |
| 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. |
| Setting | Description |
|---|---|
| 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. An announcement of the new call is played to the existing call 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 to it a base settings configuration, the phone is ready for use in Genesys Cloud. However, if you wish, 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): 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 | Disabled (Default): Allows manually setting the timezone difference. Enabled: Automatically sets the timezone for the phone based on the timezone of the DHCP server. |
| SNTP Server | Sets the name of the Simple Network Time Protocol (SNTP) server from which the phone obtains the current time. |
| GMT Offset | Sets the number of hours to offset the time of the phone from GMT. Note: When you use the Daylight Saving Time setting, 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 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 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 16384. |
| Preferred Codec List | Use the list to select and build a list of preferred media codecs in mime format. |
| DTMF Settings | |
| DTMF Payload | Specify the payload type value to use when the DTMF Method type is 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 | |
| Provision Source | Use the Provision Source list to select the source that the phone uses to obtain the provision configuration data. |
| From the Genesys Cloud provisioning service | Configure the phone to obtain provision configuration data from the Genesys Cloud provisioning service. This option will only be available if the phone supports MTLS. |
| 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 Working with Configuration Files in the Poly Documentation Library. Note: Attributes defined in the custom configuration file cannot be used to override attributes previously defined by the standard configuration file. Warning: When using the Custom Configuration Files feature, it is possible to inadvertently introduce a setting that is incompatible with Genesys Cloud which may cause the phone to malfunction. If this happens, it is the administrator's responsibility to find and remove the offending setting to restore functionality. |
| Custom Directory | Specify a corporate directory file containing contact information for people in your company. For more information, see the Corporate Directory page in the Poly Documentation Library. |
| 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. |
| DND Sync | Disabled (default): Disables Do Not Disturb (DND) synchronization between Edge and the phone. Enabled: Enables DND synchronization between Edge and the phone. |
| 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. 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.
|
| Syslog Server Address | Specify the address of the server to which you want to send the logs. |
| Syslog Port Number | 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:
|
| Application | Sets the tracing level for the Application syslog topic. |
| Configuration | Sets the tracing level for the Configuration syslog topic. |
| Micro Browser | Sets the tracing level for the Micro Browser syslog topic. |
| Copy | Sets the tracing level for the Copy syslog topic. |
| Curl | Sets the tracing level for the CURL syslog topic. |
| Key | Sets the tracing level for the Key syslog topic. |
| SIP | Sets the tracing level for the SIP syslog topic. |
| Support Objects | Sets the tracing level for the Support Objects syslog topic. |
| TLS | Sets the tracing level for the TLS syslog topic. |
| Wapp Mgr | Sets the tracing level for the Wapp Mgr syslog topic. |
| 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. |
| 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 |
|---|---|
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:
|
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:
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. |
| 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. |
| Setting | Description |
|---|---|
| Auto-Conference Settings | |
| 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 selected to use for the conference voice menu. |
| 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. |
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