Telephony Tone Parameters

Defines the maximum number of concurrent call parties that have been placed on hold to which the device can play Music on Hold (MoH) that originates from an external media player.

The maximum is 20. The default is 0.

For more information, see Configuring MoH from External Audio Source.

Note:

Enables the device to handle incoming re-INVITE messages with the "a=sendonly" attribute in the SDP, in the same way as if an "a=inactive" is received in the SDP. When enabled, the device plays a held tone to the Tel phone and responds with a SIP 200 OK containing the "a=recvonly" attribute in the SDP.

Note: The parameter is applicable only to analog interfaces.

Defines the duration (in seconds) that the dial tone is played.

Digital interfaces: The device plays the tone to an ISDN terminal. The parameter is applicable for overlap dialing if you configure the [ISDNInCallsBehavior] parameter to 65536. The dial tone is played if the ISDN Setup message doesn't include the called number. The valid range is 0 to 60. The default is 5.

Analog interfaces: The device's FXS interfaces play the dial tone after the phone is picked up (off-hook). FXO interfaces play the dial tone after the port is seized in response to ringing (from PBX/PSTN). The valid range is 0 to 60. The default time is 16.

Note:

Defines the duration (in msec) of the confirmation tone. A stutter tone is played (instead of a regular dial tone) when a Message Waiting Indication (MWI) is received. The stutter tone is composed of a confirmation tone (Tone Type #8), which is played for the defined duration (StutterToneDuration) followed by a stutter dial tone (Tone Type #15). Both these tones are defined in the CPT file.

The range is 1,000 to 60,000. The default is 2,000 (i.e., 2 seconds).

Note:

Determines whether the device plays a busy / reorder tone to the PSTN side if a Tel-to-IP call is rejected by a SIP error response (4xx, 5xx or 6xx). If a SIP error response is received, the device seizes the line (off-hook), and then plays a busy / reorder tone to the PSTN side (for the duration defined by the parameter TimeForReorderTone). After playing the tone, the line is released (on-hook).

Note: The parameter is applicable only to FXO interfaces.

Defines the duration (in seconds) of the hotline dial tone. If no digits are received during this duration, the device initiates a call to a user-defined number (configured in the Automatic Dialing table - TargetOfChannel - see Configuring Automatic Dialing).

The valid range is 0 to 60. The default is 16.

Note:

Defines the duration (in seconds) that the device plays a busy or reorder tone before releasing the line.

You can also configure this feature per specific calls, using Tel Profiles ('Time For Reorder Tone' parameter). For a detailed description of the parameter and for configuring the feature in the Tel Profiles table, see Configuring Tel Profiles.

Note: If the feature is configured for a specific Tel Profile, the device ignores this global parameter for calls associated with the Tel Profile.

Defines the delay interval (in seconds) from when the device receives a SIP BYE message (i.e., remote party terminates call) until the device starts playing a reorder tone to the FXS phone.

The valid range is 0 to 60. The default is 0.

Note: The parameter is applicable only to FXS interfaces.

Defines the duration (in seconds) of the reorder tone played to the Tel side after the IP call party releases the call, for the Cut-Through feature. After the tone stops playing, an incoming call is immediately answered if:

The valid values are 0 to 30. The default is 0 (i.e., no reorder tone is played).

Note: To enable the Cut-Through feature:

Determines whether the device plays a comfort tone (Tone Type #18) to the FXS /FXO endpoint after a SIP INVITE is sent and before a SIP 18x response is received.

Note: The parameter is applicable to FXS and FXO interfaces.

Defines the duration (in seconds) for which the offhook warning tone is played to the user.

The valid range is -1 to 2,147,483,647. The default is 600.

Note:

Enables the device to play a busy or reorder tone to the PSTN after a Tel-to-IP call is released.

Note: The parameter is applicable only to digital interfaces.

Defines an ISDN Q.8931 release cause code(s), which if mapped to the SIP release reason received from the IP side, causes the device to play a user-defined tone from the installed PRT file to the Tel side. For example, if the received SIP release cause is 480 Temporarily Unavailable and you configure the parameter with Q.931 release code 18 (No User Responding), the device plays the user-defined tone to the Tel side.

The user-defined tone is configured when creating the PRT file, using AudioCodes DConvert utility. The tone must be assigned to the "acSpecialConditionTone" (Tone Type 21) option in DConvert.

The parameter can be configured with up to 10 release codes. When configuring multiple codes, separate the codes by commas (without spaces). For example:

Q850ReasonCode2PlayUserTone = 1,18,24

If the SIP release reason received from the IP side is mapped to the Q.931 release code specified by the parameter, the device plays the user-defined tone. Otherwise, if not specified and the release code is 17 (User Busy), the device plays the busy tone and for all other release codes, the device plays the reorder tone.

Note:

Defines the playing method of the ringback tone to the Tel side.

Digital interfaces: The parameter applies to all trunks that are not configured by the [PlayRBTone2Trunk] parameter, which defines ringback tone per Trunk.

1) If the device receives a SIP 180 Ringing response (with or without SDP) and the [LocalISDNRBSource] parameter is configured to 1, it plays a ringback tone and sends an ISDN Alert with PI = 8, unless the [ProgressIndicator2ISDN_x] parameter is configured differently.
2) If the [LocalISDNRBSource] parameter is configured to 0, the device doesn't play a ringback tone and an Alert message without PI is sent to the ISDN. In this case, the PBX / PSTN plays the ringback tone to the originating terminal. Note that the receipt of a 183 response doesn't cause the device to play a ringback tone; the device sends a Progress message, unless [SIP183Behaviour] is configured to 1. If the [SIP183Behaviour] parameter is configured to 1, the 183 response is handled the same way as a 180 Ringing response.

When there are multiple 18x responses, if a 18x response without SDP is received after the remote media is played (due to a previously received 18x with SDP), the device plays the local media instead of the remote, until a new 18x with SDP is received.

1) [LocalISDNRBSource] configured to 1: The device plays a ringback tone and sends an ISDN Alert with PI = 8 to the ISDN (unless the [ProgressIndicator2ISDN_x] parameter is configured differently).

2) [LocalISDNRBSource] configured to 0: The device doesn't play a ringback tone. No PI is sent in the ISDN Alert message, unless the [ProgressIndicator2ISDN_x] parameter is configured differently. In this case, the PBX / PSTN plays a ringback tone to the originating terminal. Note that the receipt of a 183 response results in an ISDN Progress message, unless [SIP183Behaviour] is configured to 1. If [SIP183Behaviour] is configured to 1 (183 is handled the same way as a 180 with SDP), the device sends an Alert message with PI = 8 without playing a ringback tone..

Note: For ISDN trunks, this option is applicable only if the [LocalISDNRBSource] parameter is configured to 1.

Global parameter that enables the device to play a ringback tone to the IP side for IP-to-Tel calls. You can also configure this feature per specific calls, using IP Profiles ('Play RB Tone to IP' parameter). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles.

Note: If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile.

Determines whether the device plays a local ringback tone for ISDN's Two B Channel Transfer (TBCT), Release Line Trunk (RLT), or Explicit Call Transfer (ECT) call transfers to the originator when the second leg receives an ISDN Alerting or Progress message.

Note: