Voip solution using asterisk ASI

hey everyone ,
i am new to asterisk so forgive me for the silly question
i am building a voip solution , so i saw astrisk ASI can help me in this

i need guidence to build this endpoint using asterisk asi

:telephone_receiver: Create a Call
Endpoint: https://localhost.com/v1/api/create-call
Parameters:

  • to_ (required): The phone number to call.
  • from_ (required): The caller ID to display.
  • callbackurl (required): The URL to receive call status updates.
  • apikey (required): Your API key for authentication.
  • amd (optional): True/False answering machine detection (boolean).

:loud_sound: Play Audio
Endpoint: https://localhost.com/v1/api/play-audio
Parameters:

  • audiourl (required): The URL of the audio file to play.
  • uuid (required): The unique identifier of the call.

:studio_microphone: Gather Audio
Endpoint: https://localhost.com/v1/api/gather-audio
Parameters:

  • audiourl (required): The URL of the audio file to play.
  • maxdigits (required): The maximum number of digits to collect (string).
  • uuid (required): The unique identifier of the call.

:scroll: Play Text
Endpoint: https://localhost.com/v1/api/play-text
Parameters:

  • text (required): The text to convert to speech.
  • voice (required): The voice to use for text-to-speech.
  • uuid (required): The unique identifier of the call.

:studio_microphone: Gather Text
Endpoint: https://localhost.com/v1/api/gather-text
Parameters:

  • text (required): The text to convert to speech.
  • voice (required): The voice to use for text-to-speech.
  • maxdigits (required): The maximum number of digits to collect.
  • uuid (required): The unique identifier of the call.

:arrow_right_hook: Transfer Call
Endpoint: https://localhost.com/v1/api/transfer
Parameters:

  • to_ (required): The phone number to transfer the call to.
  • uuid (required): The unique identifier of the call.

:x: Hangup Call
Endpoint: https://localhost.com/v1/api/hangup
Parameters:

  • uuid (required): The unique identifier of the call.

:pause_button: Hold Call
Endpoint: https://localhost.com/v1/api/hold
Parameters:

  • uuid (required): The unique identifier of the call.

:arrow_forward: Unhold Call
Endpoint: https://localhost.com/v1/api/unhold
Parameters:

  • uuid (required): The unique identifier of the call.

:symbols: Send DTMF
Endpoint: https://localhost.com/v1/api/send-dtmf
Parameters:

  • input (required): The DTMF digits to send.
  • uuid (required): The unique identifier of the call.

:satellite: Call Status Updates
The API will send status updates to the specified callbackurl .

thanks in advance :slight_smile:

What is “asterisk ASI”? Are you asking how to build a complete telephony API solution of all those, using Asterisk?

my bad {key stroke error} i mean asterisk rest api
yes i am want to build complete telephone solution

my core motive is to :

initate call using api
play tts or .wav audio at call
capture dtmf response

ARI is fully documented, but it’s up to you to put the pieces together. I’m not sure anyone can offer anything further beyond that.

i read the document but i unable to figure out as i am new to this stufff

i wanna just do this thing

initate call using api
play tts or .wav audio at call
capture dtmf response

a lil guidance will be apricated :pray:

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