AudioCodes Mediant 1000B User Manual page 843

Analog & digital voip media gateway enterprise session border controller gateway & e-sbc
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Parameter
EMS: Blind Transfer Add Prefix
CLI: blind-xfer-add-prefix
[KeyBlindTransferAddPrefix]
EMS: Blind Transfer Disconnect
Timeout
CLI: blind-xfer-disc-tmo
[BlindTransferDisconnectTimeout]
Web: QSIG Path Replacement Mode
CLI: qsig-path-replacement-md
[QSIGPathReplacementMode]
CLI: replace-tel2ip-calnum-to
[ReplaceTel2IPCallingNumTimeout]
Version 6.8
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.
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: This parameter is applicable only to FXO and FXS
interfaces.
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 this 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.
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.
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 this 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.
Notes:
The suspension of the INVITE message occurs for all calls.
This parameter is applicable to QSIG.
843
51. Configuration Parameters Reference
Description
Mediant 1000B Gateway & SBC

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