Create base settings for the Interaction SIP Station II


Prerequisites

  • Telephony > Plugin > All permission

You can configure an Interaction SIP Station II to work in Genesys Cloud by creating a base settings profile. This profile contains a group of settings found on the Base Phone and Base Line Appearance tabs that define how a SIP station 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.

Configure the base phone

  1. Click Admin.
  2. Under Telephony, click Phone Management.
  3. Click the Base Settings tab.
  4. Click Create New and the Base Phone tab appears. ISS_Base_Phone
  5. Type a name in the Base Settings Name field.
  6. From the Phone Make and Model list, select Interaction SIP Station II.
  7. Leave Standalone Features set to Off unless you are creating a base settings configuration for conference room phones. See Enable standalone features.
  8. Perform one of the following:
  • To use the default base settings, click Save Base Settings and proceed to the Configure the base line appearance section of this article.
  • To customize the base settings, proceed to the Customize the base phone section of this article.

Customize the base phone

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

Setting

Description

Dynamic Reload Enables the dynamic reloading of the phone configuration.
Administrative Password The administrative password that you use when configuring the phone from the web interface. By default Genesys Cloud masks the password, 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.

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 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 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 decibels 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 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 values.

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
Provision Use the Provision 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). 

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.

The Message Waiting Indicator has three options:

  • None: Do not use MWI notifications.
  • Unsolicited NOTIFY: Configure the phone to receive unsolicited MWI notifications.
  • Subscription: Configure the phone to receive MWI notifications using the Subscription method. (The phone must subscribe to the MWI service to receive the notifications.)
Inter-digit Delay

Set the number of seconds to wait between the entering of each digit.

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.

Send Syslogs to Edges

Configure the phone to send data to an Edge.

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.

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 could 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 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 next to the option to enable recording of the associated item.
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. You should 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.

  1. To use your 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

  1. Click the Base Line Appearance tab.ISS_Base_Line_Appearances
  2. Type a name in the Key Label field.
  3. Perform one of the following:
  • To use the default base line appearance settings, click Save Base Settings. You can now Create an Interaction SIP Station II.
  • To customize the base line appearance settings, proceed to the Customize the base line appearance section of this article.

Customize the base line appearance

  1. In the Configuration panel, click the arrow to expand the section containing the settings you wish to customize.

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 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 Genesys Cloud automatically closes it.

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 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 If Protocol is set to UDP, which is the default setting, the UDP Settings section appears.
UDP T1 timeout Sets the amount of time, in milliseconds, before the UDP T1 connection times out.
UDP T2 timeout Sets the amount of time, in milliseconds, before the UDP T2 connection times out.

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.

  1. To use your custom base line appearance settings, click Save Base Settings.
  2. Once you have created a base line settings for the Interaction SIP Station II, you can now Create an Interaction SIP Station II.