Telephony Tone Parameters

'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:

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The parameter is applicable only to FXS interfaces.

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Each FXS port supports up to 20 concurrent MoH sessions.

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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.

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[0] Disable (default)

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[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.

Digital interfaces: The tone is played to an ISDN terminal. The parameter is applicable for overlap dialing when ISDNInCallsBehavior is set 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: 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:

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Analog interfaces: During play of dial tone, the device waits for DTMF digits.

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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:

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The parameter is applicable only to FXS interfaces.

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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.

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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).

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[0] = Disable (default)

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[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:

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The parameter is applicable only to analog interfaces.

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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.

Note: The parameter is applicable only to FXS interfaces.

'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:

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Analog interfaces: The FXS is off-hooked.

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Digital interfaces: The PSTN is connected.

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:

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CAS channels: DigitalCutThrough parameter

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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.

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[0] Disable (default)

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[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:

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A negative value indicates that the tone is played infinitely.

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The parameter is applicable only to analog interfaces.

'Play Busy Tone to Tel'

configure voip > sip-definition settings > play-bsy-tone-2tel

[PlayBusyTone2ISDN]

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

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[0] Don't Play = (Default) Immediately sends an ISDN Disconnect message.

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[1] Play when Disconnecting = Sends an ISDN Disconnect message with PI = 8 and plays a busy or reorder tone to the PSTN (depending on the release cause).

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[2] Play before Disconnect = Delays the sending of an ISDN Disconnect message for a user-defined time (configured by the TimeForReorderTone parameter) and plays a busy or reorder tone to the PSTN. This is applicable only if the call is released from the IP [Busy Here (486) or Not Found (404)] before it reaches the Connect state; otherwise, the Disconnect message is sent immediately and no tones are played.

Note: The parameter is applicable only to digital interfaces.

q850-reason-code-2play-user-tone

[Q850ReasonCode2PlayUserTone]

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 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:

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The parameter is applicable only to digital interfaces.

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To enable the feature, the 'Play Busy Tone to Tel' (PlayBusyTone2ISDN) parameter must be enabled (set to 1 or 2).

'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. Digital interfaces: The parameter applies to all trunks that are not configured by the PlayRBTone2Trunk parameter (which defines ringback tone per Trunk).

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[0] Don't Play =

✔

Ringback tone is not played.

✔

Digital Interfaces: The device doesn't play a ringback tone. No PI is sent to the PSTN unless the [ProgressIndicator2ISDN_x] parameter is configured differently.

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[1] Play on Local =

✔

Analog interfaces: Plays a ringback tone to the Tel side of the call when a SIP 180/183 response is received.

✔

Digital interfaces:
CAS: The device plays a local ringback tone to the PSTN upon receipt of a SIP 180 Ringing response (with or without SDP). Note that the receipt of a 183 response does not cause the device to play a ringback tone (unless the SIP183Behaviour parameter is set to 1).
ISDN: The device operates according to the LocalISDNRBSource parameter:
1) If the device receives a 180 Ringing response (with or without SDP) and the LocalISDNRBSource parameter is set 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 set 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 does not cause the device configured for ISDN to play a ringback tone; the device issues a Progress message (unless SIP183Behaviour is set to 1). If the SIP183Behaviour parameter is set to 1, the 183 response is handled the same way as a 180 Ringing response.

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[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.

✔

Digital interfaces: Plays according to 'Early Media'. 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 doesn't play the ringback tone; PI = 8 is sent in an ISDN Alert message (unless the [ProgressIndicator2ISDN_x] parameter is configured differently).
CAS: If a 180 response is received, but the 'early media' voice channel is not opened, the device plays a ringback tone to the PSTN.
ISDN: The device operates according to the LocalISDNRBSource parameter:
1) If LocalISDNRBSource is set 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) If LocalISDNRBSource is set 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 set to 1). If SIP183Behaviour is set to 1 (183 is handled the same way as a 180 + SDP), the device sends an Alert message with PI = 8, without playing a ringback tone.

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[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.

Note that for ISDN trunks, this option is applicable only if the LocalISDNRBSource parameter is set to 1.

'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.

'Play Local RBT on ISDN Transfer'

play-l-rbt-isdn-trsfr

[PlayRBTOnISDNTransfer]

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.

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[0] Don't Play (default)

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[1] Play

Note:

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For Blind transfer, the local ringback tone is played to first call PSTN party when the second leg receives the ISDN Alerting or Progress message.

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For Consulted transfer, the local ringback tone is played when the second leg receives ISDN Alerting or Progress message if the Progress message is received after a SIP REFER.

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The parameter is applicable only if the parameter SendISDNTransferOnConnect is set to 1.

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The parameter is applicable only to digital interfaces.

'MFC R2 Category'

mfcr2-category

[R2Category]

Defines the tone for MFC R2 calling party category (CPC). The parameter provides information on the calling party such as National or International call, Operator or Subscriber and Subscriber priority.

The value range is 1 to 15 (defining one of the MFC R2 tones). The default is 1.

Note: The parameter is applicable only to digital interfaces.