AudioCodes Mediant 800B - Page 1201

Version 7.2 1201 Mediant 800B Gateway & E-SBC
User's Manual 72. Configuration Parameters Reference
Parameter Description
[KeyBlindTransfer]
transferee destination number.
After the KeyBlindTransfer DTMF digits sequence is dialed, the
current call is put on hold (using a Re-INVITE message), a dial
tone is played to the channel, and then the phone number
collection starts.
After the destination phone number is collected, it is sent to the
transferee in a SIP REFER request in a Refer-To header. The
call is then terminated and a confirmation tone is played to the
channel. If the phone number collection fails due to a mismatch,
a reorder tone is played to the channel.
Note: For FXS/FXO interfaces, it is possible to configure
whether the KeyBlindTransfer code is added as a prefix to the
dialed destination number, by using the parameter
KeyBlindTransferAddPrefix.
blind-xfer-add-prefix
[KeyBlindTransferAddPrefix]
Determines whether the device adds the Blind Transfer code
(defined by the KeyBlindTransfer parameter) to the dialed
destination number.
 [0] Disable (default)
 [1] Enable
Note: The parameter is applicable only to analog interfaces.
blind-xfer-disc-tmo
[BlindTransferDisconnectTimeout]
Defines the duration (in milliseconds) for which the device waits
for a disconnection from the Tel side after the Blind Transfer
Code (KeyBlindTransfer) has been identified. When this timer
expires, a SIP REFER message is sent toward the IP side. If
the parameter is set to 0, the REFER message is immediately
sent.
The valid value range is 0 to 1,000,000. The default is 0.
QSIG Path Replacement Mode
qsig-path-replacement-md
[QSIGPathReplacementMode]
Enables QSIG transfer for IP-to-Tel and Tel-to-IP calls.
 [0] IP2QSIGTransfer = (Default) Enables IP-to-QSIG
transfer.
 [1] QSIG2IPTransfer = Enables QSIG-to-IP transfer.
Note: The parameter is applicable only to digital interfaces.
replace-tel2ip-calnum-to
[ReplaceTel2IPCallingNumTimeout]
Defines the maximum duration (timeout) to wait between call
Setup and Facility with Redirecting Number for replacing the
calling number (for Tel-to-IP calls).
The valid value range is 0 to 10,000 msec. The default is 0.
The interworking of the received Setup message to a SIP
INVITE is suspended when the parameter is set to any value
greater than 0. This means that the redirecting number in the
Setup message is not checked. When a subsequent Facility
with Call Transfer Complete/Update is received with a non-
empty Redirection
Number, the Calling Number is replaced with
the received redirect number in the sent INVITE message.
If the timeout expires, the device sends the INVITE without
changing the calling number.
Note:
 The suspension of the INVITE message occurs for all calls.
 The parameter is applicable only to QSIG.