Answer and Disconnect Supervision Parameters

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.

Defines a timeout (in seconds) that the device waits before sending an ISDN Release message after it has sent a Disconnect message, if no SIP message (e.g., 4xx response) is received within the timeout. The parameter is applicable when the device's trunk is configured for the Japanese NTT ISDN PRI (T1) variant (i.e., [ProtocolType] is [16], as described in Configuring Trunk Settings).

The valid value is 0 to 480. The default is 0 (i.e., timeout is 30 seconds).

For more information on this feature, see SIP-to-ISDN Disconnect Release Cause Code Mapping.

Note:

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

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.

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

The valid range is 0 to 35,791. A value of 0 means unlimited duration. The default is 0.

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

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 doesn't 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.

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Determines whether the device disconnects a call when a dial tone is detected from the PBX.

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Defines a digit pattern to send to the Tel side after a SIP 200 OK is received from the IP side. The digit pattern is a user-defined DTMF sequence that is used to indicate an answer signal (e.g., for billing).

The valid value is up to 8 characters.

Note: The parameter is applicable only to FXO and CAS.

Global parameter that defines the device's handling of calls if RTP or MSRP packets are not received within a user-defined timeout (configured by the 'Broken Connection Timeout' parameter) during an established call (i.e., packet flow suddenly stops during the call).

You can also configure this feature per specific calls, using IP Profiles ('Broken Connection Mode'). For a detailed description of the global 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.

Global parameter that defines the device's handling of calls if RTP packets are not received (detected) during early media or upon call connect (i.e., never was RTP) within a timeout. The timeout is configured by the [NoRTPDetectionTimeout] parameter.

You can also configure this feature per specific calls, using IP Profiles ('No RTP Mode' parameter). For a detailed description of the global parameter and for configuring this feature in the IP Profiles table, see Configuring IP Profiles.

Note:

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., packet 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:

Defines the timeout interval (in msec) after which a call is disconnected or re-routed if RTP packets are not received within the interval.

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

Note:

Defines the timeout (msec) for establishing MSRP connections.

The timer starts from when the device opens the MSRP media socket (port). If the timeout expires and the connection wasn't established, the device does one of the following:

The valid value is 10000 to 120000. The default is 10000 (i.e., 10 sec).

For more information, see Configuring Message Session Relay Protocol.

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, 20 for E1 and 40 for T1).

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

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

Note:

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

You can also configure the feature per specific calls, using Tel Profiles ('Disconnect Call on Detection of Busy Tone' parameter). For a detailed description of the parameter and for configuring the feature in the Tel Profiles table, see Configuring Tel Profiles.

Note:

Defines the FXO line polarity, required for DID signaling.

Typically, if the RJ-11 cabling is connected correctly (without crossing, Tip to Tip, Ring to Ring), the Tip line is positive compared to the Ring line. In this case, set the parameter to 0. With this configuration, the device assumes that the idle line polarity is Tip line positive.

When the device receives a SIP INVITE, it checks the FXO line polarity. If the polarity is "Reversed", it skips this FXO line and goes to the next line.

Note:

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

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

Note:

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

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

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Defines the voltage change slope during polarity reversal or wink.

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

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

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Defines the line voltage threshold at which a current disconnect detection is considered.

The valid range is 0 to 20 Volts. The default is 4 Volts.
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