Telephony Tone Parameters

The telephony tone parameters are described in the table below.

Tone Parameters

Parameter

Description

'Maximum simultaneous streaming calls'

max-streaming-calls

[MaxStreamingCalls]

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:

The parameter is applicable only to FXS interfaces.
Each FXS port supports up to 20 concurrent MoH sessions.
For the parameter to take effect, a device reset is required.

'SIP Hold Behavior'

configure voip > sip-definition settings > sip-hold-behavior

[SIPHoldBehavior]

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.

[0] Disable (default)
[1] Enable

Note: The parameter is applicable only to analog interfaces.

'Dial Tone Duration'

configure voip > gateway dtmf-supp-service dtmf-and-dialing > dt-duration

[TimeForDialTone]

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

Analog interfaces: 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:

Analog interfaces: During play of dial tone, the device waits for DTMF digits.
Analog interfaces: The parameter is not applicable when Automatic Dialing is enabled.

'Stutter Tone Duration'

configure voip > gateway dtmf-supp-service supp-service-settings > sttr-tone-duration

[StutterToneDuration]

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:

The parameter is applicable only to FXS interfaces.
If you want to configure the duration of the confirmation tone to longer than 16 seconds, you must increase the value of the parameter TimeForDialTone accordingly.
The MWI tone overrides the call forwarding reminder tone. For more information on MWI, see Message Waiting Indication.

'FXO AutoDial Play BusyTone'

configure voip > gateway analog fxo-setting > fxo-autodial-play-bsytn

[FXOAutoDialPlayBusyTone]

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).

[0] = Disable (default)
[1] = Enable

Note: The parameter is applicable only to FXO interfaces.

'Hotline Dial Tone Duration'

configure voip > gateway dtmf-supp-service dtmf-and-dialing > hotline-dt-dur

[HotLineToneDuration]

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:

The parameter is applicable only to analog interfaces.
You can define the Hotline duration per FXS /FXO port using the Automatic Dialing table.

'Reorder Tone Duration'

configure voip > gateway analog fxo-setting > reorder-tone-duration

[TimeForReorderTone]

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

You can also configure the feature per specific calls, using Tel Profiles (TelProfile_TimeForReorderTone). 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 settings of the global parameter is ignored for calls associated with the Tel Profile.

'Time Before Reorder Tone'

time-b4-reordr-tn

[TimeBeforeReorderTone]

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.

'Cut Through Reorder Tone Duration'

cut-thru-reord-dur

[CutThroughTimeForReOrderTone]

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:

FXS channels: CutThrough parameter

'Enable Comfort Tone'

comfort-tone

[EnableComfortTone]

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.

[0] Disable (default)
[1] Enable

Note: The parameter is applicable only yo analog interfaces.

[WarningToneDuration]

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:

A negative value indicates that the tone is played infinitely.
The parameter is applicable only to analog interfaces.

'Play Ringback Tone to Tel'

configure voip > sip-definition settings > play-rbt2tel

[PlayRBTone2Tel]

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

[0] Don't Play =
Analog Interfaces: Ringback tone is not played.
[1] Play on Local =
[2] Prefer IP = (Default):
Analog interfaces: Plays a ringback tone to the Tel side only if a 180/183 response without SDP is received. If 180/183 with SDP message is received, the device cuts through the voice channel and doesn't play the ringback tone.
[3] Play Local Until Remote Media Arrive = Plays a ringback tone according to received media. The behaviour is similar to [2]. If a SIP 180 response is received and the voice channel is already open (due to a previous 183 early media response or due to an SDP in the current 180 response), the device plays a local ringback tone if there are no prior received RTP packets. The device stops playing the local ringback tone as soon as it starts receiving RTP packets. At this stage, if the device receives additional 18x responses, it does not resume playing the local ringback tone.

'Play Ringback Tone to IP'

configure voip > sip-definition settings > play-rbt-2ip

[PlayRBTone2IP]

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 (IpProfile_PlayRBTone2IP). For a detailed description of the parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles.

Note: If this feature is configured for a specific IP Profile, the settings of this global parameter is ignored for calls associated with the IP Profile.