Answer and Disconnect Supervision Parameters

The answer and disconnect supervision parameters are described in the table below.

Answer and Disconnect Parameters

Parameter

Description

'Wait before PSTN Release-Ack'

wait-befor-pstn-rel-ack

[TimeToWaitForPstnReleaseAck]

Defines a timeout (in milliseconds) that the device waits for the receipt of an ISDN Q.931 Release message from the PSTN side before releasing the channel. The Release ACK is typically sent by the PSTN in response to the device's Disconnect message to end the call. If the timeout expires and a Release message has not yet been received, the device releases the call channel.

The valid value is 1 to 360,000. The default is 6,000.

Note: The parameter is applicable only to digital interfaces.

'Answer Supervision'

configure voip > gateway analog fxo-setting > answer-supervision

[EnableVoiceDetection]

Enables the sending of SIP 200 OK upon detection of speech, fax, or modem.

[1] Yes = The device sends a SIP 200 OK (in response to an INVITE message) when speech, fax, or modem is detected (from the Tel side, for analog interfaces).
[0] No = (Default) The device sends a SIP 200 OK only after it completes dialing (to the Tel side, for analog interfaces).

Typically, this feature is used only when early media, enabled by the [EnableEarlyMedia] parameter, is used to establish the voice path before the call is answered.

Note:

The parameter is applicable only to the Gateway application.
Digital interfaces:
To activate the feature, set the EnableDSPIPMDetectors parameter to 1.

'GW Max Call Duration'

configure voip > sip-definition settings > gw-mx-call-duration

[GWMaxCallDuration]

Defines the maximum duration (in minutes) per Gateway call. If this duration is reached, the device terminates the call. This feature is useful for ensuring available resources for new calls, by ensuring calls are properly terminated.

The valid range is 0 to 35,791, where 0 is unlimited duration. The default is 0.

Note: The parameter is applicable only to the Gateway application.

configure voip > sip-definition settings > mn-call-duration

[MinCallDuration]

Defines the minimum call duration (in seconds) for the Tel side. If an established call is terminated by the IP side before this duration expires, the device terminates the call with the IP side, but delays the termination toward the Tel side until this timeout expires.

The valid value range is 0 to 10 seconds, where 0 (default) disables this feature.

For example: assume the minimum call duration is set to 10 seconds and an IP phone hangs up a call established with a BRI phone after 2 seconds. As the call duration is less than the minimum call duration, the device does not disconnect the call on the Tel side. However, it sends a SIP 200 OK immediately upon receipt of the BYE to disconnect from the IP phone. The call is disconnected from the Tel side only when the call duration is greater than or equal to the minimum call duration.

Note:

The parameter is applicable only to the Gateway application.
The parameter is applicable to IP-to-Tel and Tel-to-IP calls.
The parameter is applicable only to ISDN protocols.

'Broken Connection Mode'

configure voip > sip-definition settings > disc-broken-conn

[DisconnectOnBrokenConnection]

Global parameter that defines the device's handling of calls if RTP packets are not received within a user-defined timeout, configured by the [BrokenConnectionEventTimeout] parameter. You can also configure this feature per specific calls, using IP Profiles (IpProfile_DisconnectOnBrokenConnection). 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.

'Broken Connection Timeout'

configure voip > sip-definition settings > broken-connection-event-timeout

[BrokenConnectionEventTimeout]

Defines the timeout interval (in 100-msec units) after which a call is disconnected if RTP packets are not received during an established call (i.e., RTP flow suddenly stops during the call).

The valid range is from 3 (i.e., 3 x 100 = 300 msec) to approx. 2684354 (i.e., 74.5 hours). The default is 100 msec.

Note:

The parameter is applicable only if the [DisconnectOnBrokenConnection] parameter is configured to [1].
Currently, the feature functions only if Silence Suppression is disabled.

configure voip > sbc settings > no-rtp-detection-timeout

[NoRTPDetectionTimeout]

Defines the timeout interval (in msec) after which a call is disconnected if RTP packets are not received within the interval. The timer begins from call setup and if no packets are received when the timer expires, the device disconnects the call.

The valid range is 0 to 50000. The default is 0, which means that this timeout feature is disabled and that the device does not disconnect the call due to RTP packets not being received.

Note:

If a call is already established and there is RTP, if at any stage during the call RTP packets are not detected for a user-defined interval, configured by [BrokenConnectionEventTimeout], the device disconnects the call, or routes it to an alternative destination, configured by the [IpProfile_DisconnectOnBrokenConnection] parameter.
The parameter is not applicable to direct media calls (see Direct Media Calls).

'Trunk Alarm Call Disconnect Timeout'

trk-alrm-call-disc-to

[TrunkAlarmCallDisconnectTimeout]

Defines the duration (in seconds) to wait after a digital trunk Red alarm (LOS / LOF) is raised, before the device disconnects the SIP call. If this timeout expires and the alarm is still raised, the device sends a SIP BYE message to terminate the call. If the alarm is cleared before this timeout expires, the call is not terminated, but continues as normal.

The range is 1 to 3600. The default is 0 (20 for BRI, ).

Note: The parameter is applicable only to the Gateway application.

'Disconnect Call on Busy Tone Detection (ISDN)'

disc-on-bsy-tone-i

[ISDNDisconnectOnBusyTone]

Determines whether a call is disconnected upon detection of a busy tone (for ISDN).

[0] Disable = (Default) Do not disconnect call upon detection of busy tone.
[1] Enable = Disconnect call upon detection of busy tone.

Note:

The parameter is applicable only to ISDN protocols.
IP-to-ISDN calls are disconnected on detection of SIT tones only in call alert state. If the call is in connected state, the SIT does not disconnect the calls. Detection of busy or reorder tones disconnects the IP-to-ISDN calls also in call connected state.

'Disconnect Call on Busy Tone Detection (CAS)'

configure voip > gateway analog fxo-setting > disc-on-bsy-tone-c

[DisconnectOnBusyTone]

Global parameter enabling call disconnection upon detection of a busy tone.

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

Note:

The parameter is applicable only to the Gateway application.
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.

Polarity (Current) Reversal for Call Release (Analog Interfaces) Parameters

'Enable Polarity Reversal'

configure voip > sip-definition settings > polarity-rvrsl

[EnableReversalPolarity]

Global parameter enabling the Line Polarity Reversal feature for call release.

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

Note:

The parameter is applicable to FXS interfaces.
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.

'Enable Current Disconnect'

configure voip > sip-definition settings > current-disc

[EnableCurrentDisconnect]

Global parameter enabling call release upon detection of a Current Disconnect signal.

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

Note:

The parameter is applicable to FXS interfaces.
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.

configure voip > interface fxs-fxo > polarity-reversal-type

[PolarityReversalType]

Defines the voltage change slope during polarity reversal or wink.

[0] = (Default) Soft reverse polarity.
[1] = Hard reverse polarity.

Note:

The parameter is applicable only to FXS interfaces.
Some Caller ID signals use reversal polarity or Wink signals, or both. In these cases, it is recommended to configure the parameter to [1] (Hard).
For the parameter to take effect, a device reset is required.

configure voip > gateway analog fxs-setting > fxs-ntt-polarity-reversal

[FXSNTTPolarityReversal]

Enables the device to comply with the NTT Japan standard for line polarity reversal for IP-to-Tel calls (FXS).

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

Note:

If this parameter is enabled, the device ignores the [EnableReversePolarity] and [TimeBeforeReorderTone] parameters for IP-to-Tel calls.
The parameter is applicable only to FXS interfaces.

configure voip > interface fxs-fxo > current-disconnect-duration

[CurrentDisconnectDuration]

Defines the duration (in msec) of the current disconnect pulse.

The range is 200 to 1500. The default is 900.

Note:

The parameter is applicable for FXS interfaces.
For the parameter to take effect, a device reset is required.