AudioCodes Mediant 500 E-SBC, Руководство пользователя

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User's Manual 852 Document #: LTRT-10437
Mediant 500 E-SBC
Parameter Description
CauseMapIsdn2Sip
[CauseMapISDN2SIP]
CauseMapISDN2SIP_SipResponse;
[\CauseMapISDN2SIP]
Release Cause Mapping from SIP to ISDN Table
Web: Release Cause Mapping
Table
EMS: SIP to ISDN Cause
Mapping
CLI: configure voip > gw
manipulations
CauseMapSip2Isdn
[CauseMapSIP2ISDN]
This table parameter maps SIP responses to Q.850 Release
Causes. The format of the ini file table parameter is as follows:
[CauseMapSIP2ISDN]
FORMAT CauseMapSIP2ISDN_Index =
CauseMapSIP2ISDN_SipResponse,
CauseMapSIP2ISDN_IsdnReleaseCause;
[\CauseMapSIP2ISDN]
54.10.8 Answer and Disconnect Supervision Parameters
The answer and disconnect supervision parameters are described in the table below.
Table 54-53: Answer and Disconnect Parameters
Parameter Description
Web: Wait before PSTN Release-
Ack
CLI: wait-befor-pstn-rel-ack
[TimeToWaitForPstnReleaseAc
k]
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: This parameter is applicable only to Digital interfaces.
Web: Answer Supervision
EMS: Enable Voice Detection
CLI: 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.

[0] No = (Default) The device sends a SIP 200 OK only after it
completes dialing.
Typically, this feature is used only when early media (enabled
using the EnableEarlyMedia parameter) is used to establish the
voice path before the call is answered.
Notes:

Digital interfaces: To activate this feature, set the
EnableDSPIPMDetectors parameter to 1.

Digital interfaces: This feature is applicable only when the
protocol type is CAS.
Web/EMS: Max Call Duration
(min)
CLI: mx-call-duration
[MaxCallDuration]
Defines the maximum duration (in minutes) of a 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. The default is 0 (i.e., no limitation).
CLI: configure voip > sip
advanced-settings > set 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