Send audio/mp3 to an established call ?

[catohagen]

As its possible to send DTMF digits to an established call, would it be possible to add sending audio/mp3 to an established call ?

something like :

service: hassio.addon_stdin
data_template:
    addon: c7744bff_ha-sip
    input:
        command: play_audio ?
        number: sip:**620@fritz.box
        audio_file: '/config/www/call_forward.mp3'

[arnonym]

I guess one could implement that. Would you like to try?

[catohagen]

if you mean by try, to test a new command if implemented, yes :) if you mean implementing command myself, I have no clue howto :/

[arnonym]

I just released a new "next" version with some new features. You can see the diff of the README here: https://github.com/arnonym/ha-plugins/compare/next?expand=1#diff-b335630551682c19a781afebcf4d07bf978fb1f8ac04c6bf87428ed5106870f5

Would be nice if you can test the changes :smile:

[catohagen]

I've just updated and tried a couple of times. With an active call i tried in developer tools / service :

service: hassio.addon_stdin
data:
    addon: 8cd50eef_ha-sip-next
    input:
        command: play_message
        number: port_04
        message: hello!
        tts_language: en

And got a loud 'beep' and log shows :

| 14:02:20.801876 [1] Playing message: hello!
| 14:02:20.812143 [ ] Error getting tts file 500 b'500: Internal Server Error'
| 14:02:22.822202 [1] Playback done.
| 14:02:22.822293 [ ] Calling webhook sip_call_webhook_id with data {'event': 'playback_done', 'sip_account': 1, 'caller': '<sip:port_04@192.168.1.161>', 'parsed_caller': 'port_04', 'type': 'message', 'message': 'hello!'}
| 14:02:22.829652 [ ] Webhook response 200 b''

But this worked :

service: hassio.addon_stdin
data:
    addon: 8cd50eef_ha-sip-next
    input:
        command: play_message
        number: port_04
        message: hello!

I get a correct single 'Hello', and spamming the 'call service' button results in , hello, hello, hello, so this works excellent :)

edit, ah...tts needs correct parameter tts_language: en-GB works :)

[catohagen]

service: hassio.addon_stdin
data:
    addon: 8cd50eef_ha-sip-next
    input:
        command: play_audio_file
        number: port_04
        audio_file: '/config/www/authentic_80s_line_noise.mp3'

This works perfect :) thank you for adding these commands..

[catohagen]

im just curious, the tts.cloud_say supports different voices :

service: tts.cloud_say
data:
  cache: false
  entity_id: media_player.living_room
  message: this is a test
  language: en-US
  options:
    voice: AmberNeural

for en-US you have these voices available :

    "en-US": [
        "JennyNeural",
        "AIGenerate1Neural",
        "AIGenerate2Neural",
        "AmberNeural",
        "AnaNeural",
        "AriaNeural",
        "AshleyNeural",
        "BrandonNeural",
        "ChristopherNeural",
        "CoraNeural",
        "DavisNeural",
        "ElizabethNeural",
        "EricNeural",
        "GuyNeural",
        "JacobNeural",
        "JaneNeural",
        "JasonNeural",
        "JennyMultilingualNeural",
        "MichelleNeural",
        "MonicaNeural",
        "NancyNeural",
        "RogerNeural",
        "SaraNeural",
        "SteffanNeural",
        "TonyNeural",
    ],

but trying this in ha-sip, the voice: parameters isnt used, its the same 'default' voice

service: hassio.addon_stdin
data:
    addon: 8cd50eef_ha-sip-next
    input:
        command: play_message
        number: port_04
        message: This is a test
        tts_language: en-US
        options:
          voice: AmberNeural

[arnonym]

Nice it's working so far!

Regarding the TTS options, ha-sip is using the REST endpoint to convert text to speach, but in the docs is no mention of options [1], so no idea if this is supported somehow.

Are you able to test the newly introduced webhooks also? If everything is working as designed, I would do a release with these changes.

[1] https://www.home-assistant.io/integrations/tts/

[catohagen]

yes, got playback_done: working sip-1-incoming.yaml:

...
answer_after: 1 # time in seconds after the call is answered (optional, defaults to 0)
webhook_to_call: # web-hook IDs which you can listen on in your actions (additional to the global web-hook)
    call_established: sip_call_established # can be all the same, or different
    entered_menu: entered_menu_webhook_id
    dtmf_digit: sip-get-dtmf-digit
    call_disconnected: sip_call_disconnected
    playback_done: sip_playback_done
    ring_timeout: sip_ring_timeout

i get the new event in logs :

| 16:47:20.818497 [1] Calling additional webhook sip_playback_done for event playback_done

but I couldn't get ring_timeout to work, or I dont see it triggered in the logs.. If I call and just leave it calling and after 60 seconds :

| 16:48:06.651390 [ ] Calling webhook sip_call_webhook_id with data {'event': 'ring_timeout', 'caller': 'sip:port_01@192.168.1.160:5060', 'parsed_caller': 'port_01', 'sip_account': 1}
| 16:48:06.659071 [ ] Webhook response 200 b''
| 16:48:06.659240 [1] Ring timeout of 60.0 triggered
| 16:48:06.659280 [1] Hang-up.
| 16:48:06.659425 [1] Call disconnected
| 16:48:06.659464 [1] Calling additional webhook sip_call_disconnected for event call_disconnected

I would assume there should be a Calling additional webhook sip_ring_timeout for event ring_timeout in the log

[arnonym]

I just tried the ring_timeout one myself, and the webhook was called.

I called a number with:

command: dial
number: sip:**123@fritz.box
ring_timeout: 3
webhook_to_call:
  ring_timeout: ring_timeout

and received the log message:

[1] Calling additional webhook ring_timeout for event ring_timeout

Not sure what you've done differently?

[catohagen]

ah...my bad, it works...i needed to add the new webhook to the automation that does the calling..

service: hassio.addon_stdin
data:
  addon: 8cd50eef_ha-sip-next
  input:
    command: dial
    number: sip:port_01@192.168.1.160:5060
    ring_timeout: 60
    sip_account: 1
    webhook_to_call:
      call_established: sip_connect_call_bridge_01
      call_disconnected: sip_call_disconnected
      ring_timeout: sip_ring_timeout

| 17:32:14.975989 [1] Registering call with id sip:port_01@192.168.1.160:5060 | 17:32:14.976038 [ ] Add to state: sip:port_01@192.168.1.160:5060 | 17:32:14.980816 [1] Calling | 17:32:15.040868 [1] Early | 17:33:14.986412 [1] Calling additional webhook sip_ring_timeout for event ring_timeout

I added it to the sip-1-incoming.yaml ...and it didnt trigger from there

This is great, I can add a mp3 playback to play when nobody picks up....or a tts message :) many possibilities now, love it...

[arnonym]

Thanks for testing!