Dynamix FXS-04 User Manual page 43

set to Inactive.
TEL NO: This field can be used as a reference remark for this line.
Normally, you can put the connected PSTN line's phone number here
for reference.
Polarity Reversal Detection: When enable the Polarity Reversal
Detection; the Dynamix FXO-04 will use the polarity reversal signal as
call was established for FXO outgoing call and start to count talking
time for Billing purpose. When disable the polarity Reversal Detection,
the Dynamix FXO-04 will use "
to set time (seconds)
Current Drop for disconnect: Use Line current drop as a disconnect
supervision to release FXO port.
Incoming call handling: The call handling policy for a FXO incoming
call.
Hot line TEL: When a FXO incoming call was detected and
after the PSTN Answer Ring Count, Dynamix FXO-04 will send
the SIP call to the specified hot line TEL number through the
Route Plan.
2 Stage Dialing: When a FXO incoming call was detected and
after the PSTN Answer Ring Count, Dynamix FXO-04 will answer
this call and play either dial tone or voice greeting file for 2
stages dialing to VOIP SIP Trunk.
Playback voice file: To enable playing voice greeting file or not. (Only
FXO use )
Repeat Count: Repeat how many counts to Play voice greeting file.
( Only FXO use )
Voice file name (MuLaw-mono 8K): Specify the file path and file
name to upload. Please make sure that the file format needs to be
G.711U, 8K, 8 bits raw file. (Only FXO use )
Flash Time: Flash Time will be sent to PSTN line (internal use only)
FAX Relay: Enable T.38 Fax Relay or T.30 Fax Bypass or not.
(T.30 Fax Bypass only supports G711a law)
Input(Encode)Gain: Adjust the volume from PSTN to VOIP (default
is 0 db)
Output(Decode)Gain: Adjust the volume from VOIP to PSTN (default
is 0 db)
Dialing Answer Delay Time (sec): When the polarity reversal
detection is disabled, Dynamix FXO-04 will answer the call(establish call
between VoIP and FXO) after time out for Billing application purpose.
After the DTMF digits dialing, Dynamix FXO-04 will send 183 with SDP to
SIP Trunk to enable the voice path for VOIP side.
PSTN Answer Ring Count: This ring count is used for called ID
detection and 2 stage dialing.
If the caller ID is sending during the first ring and second
ring, this parameter should be set to greater than or
Dialing Answer Delay Time" command
to answer the SIP call.
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