This module facilitates call transfers.
Both the prompt field and the destination number field can consist of a static or a variable value.

Destination number field can accept either a phone number (ex: 800-000-1234) or a SIP address (ex: sip:1234@

You can determine the result of the transfer using a combination of the result_variable, and the duration shadow variable (result_variable$.duration).

Transfer Result Result Variable and Duration
Caller hung up before transfer began result_variable = NULL and result_variable$.duration = NULL
Caller hung up while transfer is ringing result_variable = NULL and result_variable$.duration = 0
No one at the far end answered before the connect timeout result_variable = noanswer and result_variable$.duration = 0
Far end was busy result_variable = busy and result_variable$.duration = 0
Caller hung up during transfer result_variable = NULL and result_variable$.duration > 0
Far end hung up on the original caller result_variable = far_end_disconnect and result_variable$.duration > 0
Maximum timeout was reached result_variable = maxtime_disconnect and result_variable$.duration = maximum timeout

If the IVR is not expect to do anything after the entire transfer is finished, an Exit ( ) module should be placed at the end of the transfer module to complete the call flow.

Module Settings

Hold Music
Enable this setting to play audio while an end-user is on hold.
To upload a file for hold music, go to Application Settings > Connection Settings > Webservices > Webservice fetch audio.

This setting controls the logging function of a module.
Enabling the 'Private' setting instructs the module to not record, report, or retain the information input to that module for reporting or any other purposes. When enabled any information entered into a module during a call will be lost immediately when the call terminates.
The 'Private' setting is critical for businesses that need to maintain PCI-DSS or HIPAA compliance.

The module icon, in the upper left-hand corner, becomes grayed-out when this setting is enabled. See more details here.

Show SIP Headers
The 'Show SIP Headers' setting ONLY applies to call transfers to a SIP destination (e.g. sip:1234@ and not transfers to traditional, 10-digit phone numbers.

Enabling this option allows users to define custom headers to be sent in the event of a call transfer.
Fuse restricts custom headers to the “User-to-User” header or any header that starts with a 'x-' prefix. The field uses Fuse's standard WYSIWYG field so developers can inject dyanamic data into the headers using Fuse variables.

Show Custom Errors (Transfer)
This setting can be used to catch any error that occur during the transfer.
This setting displays 'Error' and 'No Answer' options.
If a transfer experiences a connection timeout with 'Show Custom Errors' enabled, the call automatically flows to 'No Answer'. If this setting is not enabled, the call disconnects.

Show Custom Timeouts
Enabling this setting allows users to establish custom timeout error handling for that specific module.
It will override the default global webservice timeouts in Application Settings > Connection Settings.

Enabling this presents users with two new forms in the module: Connect Timeout and Maximum Timeout.

The default durations are:
Connect Timeout: 30 seconds
Maximum Timeout: 0 (unlimited)

Advanced Settings: Advanced Fuse users may want to use shadow variables that are available for the Transfer module. For more information on this functionality, please visit the Shadow Variables page.

modules/transfer.txt · Last modified: 2020/01/09 16:04 (external edit)