Call Transfer Parameters

'Enable Transfer'

configure voip > gateway dtmf-supp-service supp-service-settings > enable-transfer

[EnableTransfer]

Enables the Call Transfer feature.

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[0] Disable

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[1] Enable = (Default)

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The device responds to a REFER message with the Referred-To header to initiate a call transfer.

Note:

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To use call transfer, the devices at both ends must support this option.

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To use call transfer, set the parameter EnableHold to 1.

'Transfer Prefix'

configure voip > gateway dtmf-supp-service supp-service-settings > transfer-prefix

[xferPrefix]

Defines the string that is added as a prefix to the transferred/forwarded called number when the REFER/3xx message is received.

Note:

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The number manipulation rules apply to the user part of the Refer-To and Contact URI before it is sent in the INVITE message.

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The parameter can be used to apply different manipulation rules to differentiate the transferred/forwarded call number from the originally dialed number.

'Transfer Prefix IP 2 Tel'

configure voip > gateway digital settings > xfer-prefix-ip2tel

[XferPrefixIP2Tel]

Defines the prefix that is added to the destination number received in the SIP Refer-To header (for IP-to-Tel calls). The parameter is applicable only to CAS blind transfer modes (i.e., LineTransferMode = 1, 2 or 3, and TrunkTransferMode = 1 or 3 for CAS).

The valid range is a string of up to 9 characters. By default, no value is defined.

Note: The parameter is also applicable to ISDN Blind Transfer, according to AT&T Toll Free Transfer Connect Service (TR 50075) “Courtesy Transfer-Human-No Data”. To support this transfer mode, you need to configure the parameter [XferPrefixIP2Tel] to "*8" (without quotation marks) and the parameter [TrunkTransferMode] to "5" (without quotation marks).

'Enable Semi-Attended Transfer'

semi-att-transfer

[EnableSemiAttendedTransfer]

Defines what the device does when a transfer is initiated while in Alerting state.

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[0] Disable = (Default) Sends a SIP REFER message with the Replaces header.

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[1] Enable = Sends a SIP CANCEL message and after a SIP 487 response is received, sends a REFER without the Replaces header.

'Blind'

configure voip > gateway analog keypad-features > blind-transfer

[KeyBlindTransfer]

Defines the keypad sequence to activate blind transfer for established Tel-to-IP calls. The Tel user can perform blind transfer by dialing the [KeyBlindTransfer] digits, followed by a 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.

configure voip > gateway digital settings > 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'

configure voip > gateway digital settings > qsig-path-replacement

[QSIGPathReplacementMode]

Enables QSIG transfer for IP-to-Tel and Tel-to-IP calls.

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[0] IP2QSIGTransfer = (Default) Enables IP-to-QSIG transfer.

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[1] QSIG2IPTransfer = Enables QSIG-to-IP transfer.

configure voip > gateway digital settings > 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:

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The suspension of the INVITE message occurs for all calls.