AudioCodes Mediant 1000 User Manual page 105

SIP User's Manual
Table 5-12: General Parameters (Advanced Parameters)
Parameter
Enable Digit Delivery to Tel
[EnableDigitDelivery]
RTP Only Mode
[RTPOnlyMode]
Version 5.2
[0] Disable = Disabled (default).
[1] Enable = Enable Digit Delivery feature for the FXO/FXS gateway
The digit delivery feature enables sending DTMF digits to the gateway's
port after the call is answered [line offhooked (FXS) or seized (FXO)].
For IP-to-Tel calls, after answering the call, the gateway plays the DTMF
digits (of the called number) towards the phone line.
For digital modules: If the called number in IP-to-Tel call includes the
characters 'w' or 'p', the gateway places a call with the first part of the
called number (before 'w' or 'p') , and plays DTMF digits after the call is
answered. If the character 'w' is used, the gateway waits for detection of
dial tone before it starts playing DTMF digits. For example, if the number
'1007766p100' is defined as the called number, the gateway places a
call with 1007766 as the destination number, then, after the call is
answered, it waits for 1.5 seconds and plays the rest of the number
(100) as DTMF digits.
Other examples: 1664wpp102, 66644ppp503, 7774w100pp200.
Notes:
The called number can include characters 'p' (1.5 seconds pause)
and 'd' (detection of dial tone). If character 'd' is used, it must be the
first 'digit' in the called number. The character 'p' can be used several
times.
For example (for FXS/FXO module), the called number can be as
follows: d1005, dpp699, p9p300. To add the 'd' and 'p' digits, use the
usual number manipulation rules.
To use this feature with FXO modules, configure the gateway to
operate in one stage dialing mode.
If the parameter EnableDigitDelivery is enabled, it is possible to
configure the FXS/FXO gateway to wait for dial tone per destination
phone number (before or during dialing of destination phone number),
therefore, the parameter IsWaitForDialTone (configurable for the
entire gateway) is ignored.
The FXS and digital modules send SIP 200 OK responses only after
the DTMF dialing has completed.
Enables the gateway to start sending and/or receiving RTP packets to
and from remote endpoints without the need to establish a Control
session. The remote IP address is determined according to the Tel to IP
Routing table. The port is the same port as the local RTP port (set by
BaseUDPPort and the channel on which the call was received).
Valid options include:
[0] Disable = Disable (default).
[1] Transmit & Receive = send and receive RTP.
[2] Transmit Only= send RTP only.
[3] Receive Only= receive RTP only.
105
5. Web-based Management
Description
September 2007