AudioCodes Mediapack mp-11x, User Manual

Page: 103 / 390

Version 5.2 103 August 2007
SIP User's Manual 5. Web-based Management
Table 5-12: General Parameters (Advanced Parameters)
Parameter Description
Reanswer Time
[RegretTime]
The time period after user hangs up the phone and before the call is
disconnected (FXS). Also called regret time.
The valid range is 0 to 255 (in seconds). The default value is 0.
Disconnect and Answer Supervision
Enable Polarity Reversal
[EnableReversalPolarity]
ƒ
[0]
Disable = Disable the polarity reversal service (default).
ƒ
[1]
Enable = Enable the polarity reversal service.
If the polarity reversal service is enabled, then the FXS gateway
changes the line polarity on call answer and changes it back on call
release.
The FXO gateway sends a 200 OK response when polarity reversal
signal is detected (applicable to one stage dialing only), and releases a
call when a second polarity reversal signal is detected.
Enable Current Disconnect
[EnableCurrentDisconnec
t]
ƒ
[0]
Disable = Disable the current disconnect service (default).
ƒ
[1]
Enable = Enable the current disconnect service.
If the current disconnect service is enabled, the FXO releases a call
when current disconnect signal is detected on its port, while the FXS
gateway generates a 'Current Disconnect Pulse' after a call is released
from IP.
The current disconnect duration is determined by the parameter
CurrentDisconnectDuration. The current disconnect threshold (FXO
only) is determined by the parameter
CurrentDisconnectDefaultThreshold. The frequency at which the analog
line voltage is sampled is determined by the parameter
TimeToSampleAnalogLineVoltage.
Disconnect on Broken
Connection
[DisconnectOnBrokenCo
nnection]
ƒ
[0]
No = Don't release the call.
ƒ
[1]
Yes = Call is released if RTP packets are not received for a
predefined timeout (default).
Notes:
ƒ
If enabled, the timeout is set by the parameter
BrokenConnectionEventTimeout, in 100 msec resolution. The default
timeout is 10 seconds (BrokenConnectionEventTimeout =100).
ƒ
This feature is applicable only if RTP session is used without Silence
Compression. If Silence Compression is enabled, the gateway
doesn't detect that the RTP connection is broken.
ƒ
During a call, if the source IP address (from where the RTP packets
are sent) is changed without notifying the gateway, the gateway filters
these RTP packets. To overcome this issue, set
DisconnectOnBrokenConnection = 0; the gateway doesn't detect
RTP packets arriving from the original source IP address, and
switches (after 300 msec) to the RTP packets arriving from the new
source IP address.