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'

configure voip > gateway digital settings > 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.

configure voip > interface e1-t1 > isdn-japan-ntt-timer-t305

[ISDNJapanNttTimerT305]

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:

For the parameter to take effect, a device restart is required.

'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 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.

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 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.

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 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.

'No RTP Mode'

configure voip > sbc settings > no-rtp-mode

[NoRTPMode]

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:

If you configure this feature for a specific IP Profile, the device ignores this global parameter for calls associated with the IP Profile.
The parameter is applicable only to the SBC application.

'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., 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:

The parameter is applicable only when the 'Broken Connection Mode' parameter is configured to Disconnect.
For this feature to function, you also need to disable silence suppression.

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

[NoRTPDetectionTimeout]

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:

The parameter is applicable to the 'Broken Connection Mode' and 'No RTP Mode' parameters.
If a call is already established and RTP exists and at any stage during the call packets are not detected for a user-defined interval (configured by the 'Broken Connection Timeout' parameter), the device disconnects the call or routes it to an alternative destination, configured by the IP Profile parameter 'Broken Connection Mode'.
The parameter is not applicable to MSRP and direct media calls (see Direct Media Calls).

'Timeout to Establish MSRP Connection'

configure voip > sbc settings > msrp-connection-establish-timeout

[MSRPConnectionEstablishTimeout]

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:

Ends the SIP session.
If you've enabled the Broken Connection feature, the device searches for an alternative route in the IP-to-IP Routing table. For more information, see the IP Profile's 'Broken Connection Mode' parameter in Configuring IP Profiles.

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

For more information, see Configuring Message Session Relay Protocol.

'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 E1 and 40 for T1).

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

'Disconnect Call on Busy Tone Detection (ISDN)'

configure voip > gateway digital settings > 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 doesn't disconnect the calls. Detection of busy or reorder tones disconnects the IP-to-ISDN calls also in call connected state.