Answer and Disconnect Supervision Parameters
The answer and disconnect supervision parameters are described in the table below.
Answer and Disconnect Parameters
Parameter |
Description |
||||||||||||||||||
---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
'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.
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:
|
||||||||||||||||||
'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. |
||||||||||||||||||
'Disconnect on Dial Tone' configure voip > gateway analog fxo-setting > disc-on-dial-tone [DisconnectOnDialTone] |
Determines whether the device disconnects a call when a dial tone is detected from the PBX.
Note:
|
||||||||||||||||||
'Send Digit Pattern on Connect' configure voip > sip-definition settings > digit-pttrn-on-conn [TelConnectCode] |
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. |
||||||||||||||||||
'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:
|
||||||||||||||||||
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:
|
||||||||||||||||||
'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:
|
||||||||||||||||||
Polarity (Current) Reversal for Call Release (Analog Interfaces) Parameters |
|||||||||||||||||||
configure voip > interface fxs-fxo > default-linepolarity-state [SetDefaultLinePolarityState] |
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:
|
||||||||||||||||||
'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:
|
||||||||||||||||||
'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:
|
||||||||||||||||||
configure voip > interface fxs-fxo > polarity-reversal-type [PolarityReversalType] |
Defines the voltage change slope during polarity reversal or wink.
Note:
|
||||||||||||||||||
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).
Note:
|
||||||||||||||||||
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:
|
||||||||||||||||||
[CurrentDisconnectDefaultThreshold] |
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.
|
||||||||||||||||||
configure voip > interface fxs-fxo > time-to-sample-analog-line-voltage [TimeToSampleAnalogLineVoltage] |
Defines the frequency at which the analog line voltage is sampled (after offhook), for detection of the current disconnect threshold. The valid range is 100 to 2500 msec. The default is 1000 msec. Note:
|