Transfer

Voice Gateway Transfer Node

Description

The Transfer Node allows you to transfer an ongoing call to a Tel or SIP target.

Settings

General settings

Parameter	Type	Description	Transfer Type
Transfer Type	Dropdown	There are two transfer types: - Refer — forwarding an existing call. - Dial — creating a new outgoing call. If you want to use this type and still have the old Node version, add a new Voice Gateway Transfer Node in the Flow Editor and manually transfer the required settings from the old Node.	-
Reason	CognigyScript	The reason for the handover. It is shown in Voice Gateway logs.	All
Target	CognigyScript	E.164 syntax or a SIP URI are supported.	All
Caller ID	Number	The caller ID. Some carriers, like Twilio, require a registered number for outgoing calls.	Dial
Dial Music	URL	Custom audio or ring-back which plays to the caller while the outbound call is ringing. Only the .wav or .mp3 formats are supported.	Dial
Timeout	Number	The amount of time (in seconds) that the AI Agent will ring before a no-answer timeout. The default value is 60 seconds.	Dial
Enable Copilot	Toggle	Creates the UUIValue which will be sent to the Contact Center through SIP Headers. Will collect information from the Transcription Webhook field as well as from the VG Endpoint Copilot Config field to create the UUIValue. This setting requires a configured Voice Copilot Endpoint.	Dial
Copilot Headers Key	CognigyScript	Define the Copilot Headers Key. The key is provider-dependent. For example, for Genesys, the key is User-to-User. The generated value becomes accessible after engaging the Transfer Node, located in the input object as {{input.UUIValue}}. Note that you do not need to handle this value manually. It is automatically generated and added to the header.	Dial

Transcribe

Parameter	Type	Description	Transfer Type
STT Vendor	Selector	This option is only available on Voice Gateway. Select the desired STT Vendor. For custom use all lowercase letters like microsoft, google, aws, or nuance.	Dial
STT Language	Selector	Select the desired STT Language. For custom languages, use the following format: de-DE, fr-FR, en-US.	Dial
STT Label	CognigyScript	The alternative name of the vendor is the one you specify in the Voice Gateway Self-Service Portal. If you have created multiple speech services from the same vendor, use the label to specify which service to use.	Dial
Disable STT Punctuation	Toggle	This parameter is active only when Google or Deepgram is selected in the STT Vendor setting. Prevents the STT response from the AI Agent to include punctuation marks.	Dial
Deepgram Model	Selector	This parameter is active only when Deepgram is selected in the STT Vendor setting. Choose a model for processing submitted audio. Each model is associated with a tier. Ensure that the selected tier is available for the chosen STT language. For detailed information about Deepgram models, refer to the Deepgram documentation.	Dial
Endpointing	Toggle	This parameter is active only when Deepgram is selected in the STT Vendor setting. Deepgram's Endpointing feature watches streaming audio for long pauses that signal the end of speech. When it spots an endpoint, it finalizes predictions and returns the transcript, marking it as complete with the speech_final parameter set to true. For detailed information about Deepgram Endpointing, refer to the Deepgram documentation. The duration for detecting the end of speech is preconfigured with a default value (10 milliseconds). If you want to change this value, use the Endpointing Time setting.	Dial
Endpointing Time	Number	This parameter is active only when Deepgram is selected in the STT Vendor setting and the Endpointing toggle is enabled. Customize the duration (in milliseconds) for detecting the end of speech. The default is 10 milliseconds of silence. Transcripts are sent after detecting silence, and the system waits until the speaker resumes or the required silence time is reached. Once either condition is met, a transcript is sent back with speech_final set to true.	Dial
Smart Formatting	Toggle	This parameter is active only when Deepgram is selected in the STT Vendor setting. Deepgram's Smart Format feature applies additional formatting to transcripts to optimize them for human readability. Smart Format capabilities vary between models. When Smart Formatting is turned on, Deepgram will always apply the best-available formatting for your chosen model, tier, and language combination. For detailed examples, refer to the Deepgram documentation. Note that when Smart Formatting is turned on, punctuation will be activated, even if you have the Disable STT Punctuation setting enabled.	Dial
Google Model	Dropdown	This parameter is active only when Google is selected in the STT Vendor setting. Utilizes one of Google Cloud Speech-to-Text transcription models, with the latest_short model being the default choice. For a detailed list of Google models, refer to the Transcription models section in the Google Documentation. Keep in mind that the default value is a Google Model type that can be used if other models don't suit your specific scenario.	Dial
Transcription Webhook	URL	The Webhook is triggered with an HTTP POST whenever an interim or final transcription is received. Uses the default recognizer. Note that if the Voice Copilot Endpoint is inactive, you can use any Webhook URL to receive voice call transcripts. However, when the Voice Copilot Endpoint is enabled, ensure that the specified Webhook URL is associated with it for processing.	Dial
Audio Stream Selection	Selector	Select the source of the audio stream: - Caller/Called - both the incoming and outgoing audio streams of the caller and the called party. - Caller - the incoming and outgoing audio stream of the caller. - Called - the incoming and outgoing audio stream of the called party. Ensure that the selected audio stream matches the language specified for transcription. If no audio stream is provided, the system will use the one set in the beginning, which should also match the language specified for transcription.	Dial

Answering Machine Detection

Answering Machine Detection (AMD) determines whether an outgoing call has been answered by a human or a machine (such as a voicemail or answering machine). When AMD is enabled, events are generated in the Flow to indicate the detection result, allowing you to take specific actions based on the outcome.

For example, if a call is transferred to a user and AMD detects a machine, you will receive an event indicating the machine detection.

Parameter	Type	Description	Transfer Type
Answering Machine Detection	Toggle	Indicates whether a call has been answered by a human agent or a machine.	Dial
Redirect on machine detected	Selector	Redirects the call back to the Flow in case a machine was detected.	Dial
Redirect text	Text	Text to say after machine was detected before going back to the Flow. For example, Mailbox detected, your call will be redirected back to the AI Agent. Please stay in the call.	Dial

Advanced

Parameter	Type	Description	Transfer Type
Custom Transfer SIP Headers	Toggle	Data that needs to be sent as SIP headers in the generated SIP message.	All
Transfer SIP Headers	JSON	Array of objects listing SIP headers that should be sent to the transferee. Each object comprises a name and a value attribute.	All
Referred By	String	This parameter is optional. This setting allows you to change the original Referred By value, which can be a SIP URI or a user identifier such as a phone number. To define the Referred By value, you can use the following patterns: - SIP URI - sip:[referred-by]@custom.domain.com. In this case, the entire SIP URI will be sent as the Referred-By header. Example: "Referred-by": "sip:CognigyOutbound@custom.domain.com". - User Identifier - sip:[referred-by]@[SIP FROM Domain from carrier config]. Example, "Referred-By": "sip:CognigyOutbound@sip.cognigy.ai".	Refer

Troubleshooting

Possible 403 Error After Dial Transfer

In case you receive a 403 error log message after using a Transfer Node with Dial, check the related PCAP file in the Voice Gateway Self-Service Portal first. There you should get more information about the reason for the 403 error.

Some SBCs like Twilio may reject calls from unknown numbers. In this case, it might be necessary to configure the P-Asserted-Identity header for the Transfer. To do this, turn on the Custom Transfer SIP Headers toggle in the Advanced settings of the Transfer Node, and enter the following in the JSON field:

 {
 "P-Asserted-Identity": "<sip:phonenumber@host>"
 }