ha-sip

Home Assistant SIP/VoIP Gateway is a Home Assistant add-on which

allows the dialing and hanging up of phone numbers through a SIP end-point
triggering of services through dial tones (DTMF) after the call was established.
listens for incoming calls and can trigger actions through a web-hook (the call is not picked up)
accepting calls (optionally filtered by number)
handle PIN input before triggering actions
send DTMF digits to an established call (incoming or outgoing)
speak to Home Assistant Voice Assist without special hardware

Installation

This add-on is for the Home Assistant OS or supervised installation methods mentioned in https://www.home-assistant.io/installation/. With that in place you can install this third-party plug-in like described in https://www.home-assistant.io/common-tasks/os#installing-third-party-add-ons. The repository URL is https://github.com/arnonym/ha-plugins.

Note: Alternatively you can run ha-sip in a stand-alone mode (for Home Assistant Container installations). In that mode the communication to ha-sip will be handled by MQTT. You can find the installation steps at the end of this document.

After that you need to configure your SIP account(s), TTS parameters and webhook ID. The default configuration looks like this:

sip_global:
    port: 5060
    log_level: 5 # log level of pjsip library
    name_server: '' # comma separated list of name servers, must be set if sip server must be resolved via SRV record
    cache_dir: '/config/audio_cache' # directory to cache TTS messages or converted audio files. Must be inside /config and existing
    global_options: ''
sip:
    enabled: true
    registrar_uri: sip:fritz.box
    id_uri: sip:[email protected]
    realm: '*'
    user_name: homeassistant
    password: secure
    answer_mode: listen  # "listen" or "accept", see below
    settle_time: 1 # time to wait for playing the message/actions/etc. after call was established
    incoming_call_file: "" # config and menu definition file for incoming calls, see below
    options: ''
sip_2:
    enabled: false
    registrar_uri: sip:fritz.box
    id_uri: sip:[email protected]
    realm: '*'
    user_name: anotheruser
    password: secret
    answer_mode: listen
    settle_time: 1
    incoming_call_file: ""
    options: ''
tts:
    engine_id: tts.google_translate_de_com # entity id of the TTS engine
    platform: google_translate # deprecated, must not be set if engine_id is set
    language: en # might also be in en-US format, depending on the platform
    debug_print: false # set to true, to output known engines and languages to the log at startup
    voice: zephyr # voice if engine supports it
webhook:
    id: sip_call_webhook_id

Note: When your user_name or password starts with a number, you need to put it in quotes like "1234".

Note For TTS you need to install one of the TTS integrations. If you're unsure about the entity id used for engine_id, set debug_print to true and restart the add-on. The add-on will output a list of all available engines and languages into the log. If the configured engine and language is valid, it will also log the available voices (if the engine supports it).

For `global_options` you can specify the following options

  --stun-server STUN_SERVER
                        STUN server to use for NAT traversal (default: None)
  --udp {enabled,enable,true,yes,on,1,disabled,disable,false,no,off,0}
                        Enable or disable UDP transport (default: enabled)
  --tcp {enabled,enable,true,yes,on,1,disabled,disable,false,no,off,0}
                        Enable or disable TCP transport (default: enabled)
  --tls {enabled,enable,true,yes,on,1,disabled,disable,false,no,off,0}
                        Enable or disable TLS transport (default: disabled)
  --tls-port TLS_PORT   Port to use for TLS transport (default: 5061)

For `options` on each SIP account there are

  --proxy PROXY         Proxy server to use for SIP (default: None)
  --ice {enabled,enable,true,yes,on,1,disabled,disable,false,no,off,0}
                        Enable or disable ICE (default: true)
  --use-stun-for-sip {enabled,enable,true,yes,on,1,disabled,disable,false,no,off,0}
                        Enable or disable STUN for sip (default: true)
  --use-stun-for-media {enabled,enable,true,yes,on,1,disabled,disable,false,no,off,0}
                        Enable or disable STUN for media (default: true)
  --use-contact-rewrite {enabled,enable,true,yes,on,1,disabled,disable,false,no,off,0}
                        Enable or disable contact rewrite for SIP (default: true)
  --use-via-rewrite {enabled,enable,true,yes,on,1,disabled,disable,false,no,off,0}
                        Enable or disable via rewrite for SIP (default: true)
  --use-sdp-nat-rewrite {enabled,enable,true,yes,on,1,disabled,disable,false,no,off,0}
                        Enable or disable SDP NAT rewrite for SIP (default: true)
  --use-sip-outbound {enabled,enable,true,yes,on,1,disabled,disable,false,no,off,0}
                        Enable or disable SIP outbound (default: true)
  --turn-server TURN_SERVER
                        Set the TURN server to use for SIP (default: None)
  --turn-connection-type {tcp,udp,tls}
                        Set the TURN server connection protocol (default: udp)
  --turn-user TURN_USER
                        Set the TURN user (default: None)
  --turn-password TURN_PASSWORD
                        Set the TURN password (default: None)

Usage

Outgoing calls

Outgoing calls are made via the hassio.addon_stdin service in the action part of an automation. To be able to enter the full command, you must switch to YAML mode by clicking on the menu with the triple dot and selecting Edit in YAML.

You can use dial and hangup with the hassio.addon_stdin service to control outgoing calls in an action in your automation:

service: hassio.addon_stdin
data:
    addon: c7744bff_ha-sip
    input:
        command: dial
        number: sip:**[email protected] # number to call. Format depends on your SIP provider, 
                                    # but might look like 'sip:[email protected]' for external calls
        webhook_to_call: # web-hook IDs which you can listen on in your actions (additional to the global web-hook)
            ring_timeout: another_webhook_id # can be all the same, or different
            call_established: another_webhook_id
            entered_menu: another_webhook_id
            timeout: another_webhook_id  # is called after the given time-out on a menu is reached
            dtmf_digit: another_webhook_id # is called when the calling party sends a DTMF tone
            call_disconnected: another_webhook_id
            playback_done: another_webhook_id # is called after playback of message or audio file is done
        ring_timeout: 15 # time to ring in seconds (optional, defaults to 300)
        sip_account: 1 # number of configured sip account: 1, 2, or 3 
                       # (optional, defaults to first enabled sip account)
        menu:
            message: There's a burglar in da house.

If there is already an outgoing call to the same number active, the request will be ignored.

To hang up the call again:

service: hassio.addon_stdin
data:
    addon: c7744bff_ha-sip
    input:
        command: hangup
        number: sip:**[email protected]

To send DTMF digits to an established call:

service: hassio.addon_stdin
data:
    addon: c7744bff_ha-sip
    input:
        command: send_dtmf
        number: sip:**[email protected]
        digits: "123#"
        method: in_band # method can be "in_band" (default), "rfc2833" or "sip_info"

Note: When using a # digit, you need to put the whole sequence in quotes, eg. "#5".

Warning You can't use the post_action with send_dtmf because I don't see a way to know when PJSIP is done sending the tones.

To transfer a call to a different SIP URI:

service: hassio.addon_stdin
data:
    addon: c7744bff_ha-sip
    input:
        command: transfer
        number: sip:**[email protected]
        transfer_to: sip:**[email protected]

To bridge the audio streams of two active calls:

service: hassio.addon_stdin
data:
    addon: c7744bff_ha-sip
    input:
        command: bridge_audio
        number: sip:**[email protected]
        bridge_to: sip:**[email protected]

To play a message through TTS

service: hassio.addon_stdin
data:
    addon: c7744bff_ha-sip
    input:
        command: play_message
        number: sip:**[email protected]
        message: hello!
        tts_language: en
        cache_audio: true # If message should be cached in `cache_dir`. 
                          # Defaults to false. `cache_dir` must be configured in ha-sip config.
                          # Don't enable this for dynamic messages, you'll just fill your storage.
        wait_for_audio_to_finish: true # Do not accept DTMF tones until the message has been played

To play an audio file

service: hassio.addon_stdin
data:
    addon: c7744bff_ha-sip
    input:
        command: play_audio_file
        number: sip:**[email protected]
        audio_file: '/config/audio/welcome.mp3'
        cache_audio: true # If converted file should be cached in `cache_dir`. 
                          # Defaults to false. `cache_dir` must be configured in ha-sip config
        wait_for_audio_to_finish: true # Do not accept DTMF tones until the audio file has been played

To stop audio playback (both audio file and message):

service: hassio.addon_stdin
data:
    addon: c7744bff_ha-sip
    input:
        command: stop_playback
        number: sip:**[email protected]

Incoming calls

Listen mode

In listen mode no call will be answered (picked up) but you can trigger an automation through a Webhook trigger for every incoming call. The webhook ID must match the ID set in the configuration.

You can get the caller from {{trigger.json.caller}} or {{trigger.json.parsed_caller}} for usage in e.g. the action of your automation. If you also use the menu ID webhook you also need to check for {{ trigger.json.event == "incoming_call" }} e.g. in a "Choose" action type.

Example of "incoming call" webhook message:

{
    "event": "incoming_call",
    "caller": "<sip:[email protected]>",
    "parsed_caller": "5551234456",
    "sip_account": 1
}

You can also answer an incoming call from home assistant by using the hassio.addon_stdin service:

service: hassio.addon_stdin
data:
    addon: c7744bff_ha-sip
    input:
        command: answer
        number: "5551234456" # if this is unclear, you can look that up in the logs ("Registering call with id <number>")
        webhook_to_call: # optional web-hook IDs which you can listen on in your actions (additional to the global web-hook)
            call_established: another_webhook_id
            entered_menu: another_webhook_id
            timeout: another_webhook_id  # is called after the given time-out on a menu is reached
            dtmf_digit: another_webhook_id # is called when the calling party sends a DTMF tone
            call_disconnected: another_webhook_id
            playback_done: another_webhook_id # is called after playback of message or audio file is done
        menu:
          message: Bye
          post_action: hangup

If you don't provide a menu the menu from incoming_call_file will be used.

Accept mode

In accept mode you can additionally make ha-sip to accept the call. For this you can define a menu per SIP account. Put a config file into your /config directory of your home-assistant installation (e.g. use the samba add-on to access that directory).

Example content of /config/sip-1-incoming.yaml:

allowed_numbers: # list of numbers which will be answered. If removed all numbers will be accepted
    - "5551234456"
    - "5559876543"
    - "555{*}" # matches every number starting with 555
    - "555{?}" # matches every number starting with 555 which is 4 digits long
# blocked_numbers: # alternatively you can specify the numbers not to be answered. You can't have both.
#    - "5551234456"
#    - "5559876543"
answer_after: 0 # 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: another_webhook_id # can be all the same, or different
    entered_menu: another_webhook_id
    dtmf_digit: another_webhook_id
    call_disconnected: another_webhook_id
menu:
    message: Please enter your access code
    choices_are_pin: true
    choices:
        '1234':
            id: owner
            message: Welcome beautiful.
            post_action: hangup
        '5432':
            id: maintenance
            message: Your entrance has been logged.
            post_action: hangup
        'default':
            id: wrong_code
            message: Wrong code, please try again
            post_action: return

After that you set incoming_call_file in the add-on configuration to /config/sip-1-incoming.yaml.

Call menu definition

used for incoming and outgoing calls.

menu:
    id: main # If "id" is present, a message will be sent via webhook (entered_menu), see below (optional)
    message: Please enter your access code # the message to be played via TTS (optional, defaults to empty)
    language: en # TTS language (optional, defaults to the global language from add-on config)
    choices_are_pin: true # If the choices should be handled like PINs (optional, defaults to false)
    timeout: 10 # time in seconds before "timeout" choice is triggered (optional, defaults to 300)
    post_action: noop # this action will be triggered after the message was played. Can be 
                      # "noop" (do nothing), 
                      # "return <level>" (makes only sense in a sub-menu, returns <level> levels, defaults to 1), 
                      # "hangup" (hang-up the call) and
                      # "repeat_message" (repeat the message until the time-out is reached)
                      # "jump <menu-id>" (jumps to menu with id <menu-id>)
                      # (optional, defaults to noop)
    action: # action to run when menu was entered (before playing the message) (optional)
        # For details visit https://developers.home-assistant.io/docs/api/rest/, POST on /api/services/<domain>/<service>
        domain: switch # home-assistant domain
        service: turn_on # home-assistant service
        entity_id: switch.open_front_door # home assistant entity
    choices: # the list of actions available through DTMF (optional)
        '1234': # DTMF sequence, and definition of a sub-menu
            id: owner # same as above, also any other option from above can be used in this sub-menu
            message: Welcome beautiful.
            cache_audio: true # If message should be cached in `cache_dir`. 
                              # Defaults to false. `cache_dir` must be configured in ha-sip config.
                              # Don't enable this for dynamic messages, you'll just fill your storage.
            wait_for_audio_to_finish: true # Do not accept DTMF tones until the message/audio file has been played
            post_action: hangup 
        '5432':
            id: maintenance
            message: Your entrance has been logged.
            post_action: hangup
        '7777':
            audio_file: '/config/audio/welcome.mp3' # audio file to be played (.wav or .mp3).
            post_action: jump owner # jump to menu id 'owner'
        'default': # this will be triggered if the input does not match any specified choice
            id: wrong_code
            message: Wrong code, please try again
            post_action: return
        'timeout': # this will be triggered when there is no input 
            id: timeout
            message: Bye.
            post_action: hangup

Note: The audio files need to reside in your home-assistant config directory, as this is the only directory accessible inside the add-on.

Web-hooks

For most events in ha-sip there's a web-hook triggered. The property internal_id is the number you can use to identify the call in your automations.

`incoming_call`

{
    "event": "incoming_call",
    "caller": "<sip:[email protected]>",
    "parsed_caller": "5551234456",
    "sip_account": 1,
    "internal_id": "something-unique"
}

`call_established`

{
    "event": "call_established",
    "caller": "<sip:[email protected]>",
    "parsed_caller": "5551234456",
    "sip_account": 1,
    "internal_id": "something-unique"
}

`entered_menu`

{
    "event": "entered_menu",
    "caller": "<sip:[email protected]>",
    "parsed_caller": "5551234456",
    "menu_id": "owner",
    "sip_account": 1,
    "internal_id": "something-unique"
}

`dtmf_digit`

{
    "event": "dtmf_digit",
    "caller": "<sip:[email protected]>",
    "parsed_caller": "5551234456",
    "digit": "1",
    "sip_account": 1,
    "internal_id": "something-unique"
}

`call_disconnected`

{
    "event": "call_disconnected",
    "caller": "<sip:[email protected]>",
    "parsed_caller": "5551234456",
    "sip_account": 1,
    "internal_id": "something-unique"
}

`playback_done` for message (TTS)

{
    "event": "playback_done",
    "caller": "<sip:[email protected]>",
    "parsed_caller": "5551234456",
    "sip_account": 1,
    "type": "message",
    "message": "message that has been played",
    "internal_id": "something-unique"
}

`playback_done` for audio file

{
    "event": "playback_done",
    "caller": "<sip:[email protected]>",
    "parsed_caller": "5551234456",
    "sip_account": 1,
    "type": "audio_file",
    "audio_file": "/config/audio/welcome.mp3",
    "internal_id": "something-unique"
}

`ring_timeout`

{
    "event": "ring_timeout",
    "caller": "<sip:[email protected]>",
    "parsed_caller": "5551234456",
    "sip_account": 1,
    "internal_id": "something-unique"
}

`timeout`

{
    "event": "timeout",
    "caller": "<sip:[email protected]>",
    "parsed_caller": "5551234456",
    "sip_account": 1,
    "menu_id": "main",
    "internal_id": "something-unique"
}

Examples

Trigger services through DTMF on an outgoing call

service: hassio.addon_stdin
data:
    addon: c7744bff_ha-sip
    input:
        command: dial
        number: sip:**[email protected]
        menu:
            message: Press one to open the door, two to turn on light outside, three to play music
            choices:
                '1':
                    message: Door has been opened
                    action:
                        domain: switch
                        service: turn_on
                        entity_id: switch.open_front_door
                '2':
                    message: Light outside has been switched on
                    action:
                        domain: light
                        service: turn_on
                        entity_id: light.outside
                '3':
                    message: Play music
                    action:
                        domain: script
                        service: turn_on
                        entity_id: script.play_music_please
                        service_data:
                          variables:
                            song: 'Never gonna give you up'

Play a message without DTMF interaction on sip account 1

service: hassio.addon_stdin
data:
    addon: c7744bff_ha-sip
    input:
        command: dial
        number: sip:**[email protected]
        ring_timeout: 15
        sip_account: 1
    menu:
        message: There's a burglar in da house.

Use PIN protection on outgoing call

service: hassio.addon_stdin
data:
    addon: c7744bff_ha-sip
    input:
        command: dial
        number: sip:**[email protected]
        menu:
            message: Please enter your access code
            choices_are_pin: true
            timeout: 10
            choices:
                '1234':
                    id: owner
                    message: Welcome beautiful.
                    post_action: hangup
                '5432':
                    id: maintenance
                    message: Your entrance has been logged.
                    post_action: hangup
                'default':
                    id: wrong_code
                    message: Wrong code, please try again
                    post_action: return
                'timeout':
                    id: timeout
                    message: Bye.
                    post_action: hangup

All the examples are working also for incoming calls when you copy the menu part into your incoming configuration yaml.

Troubleshooting

The first place to look is the log of the ha-sip add-on. There you can see individual SIP messages and the logs of ha-sip itself (prefixed with "|").

Stand-alone mode

The stand-alone mode can be used if you run home assistant in a docker environment and you don't have access to the hassio.addon_stdin service. Instead of stdin - MQTT will be used for communication.

Follow the instructions from home assistant to set up a working MQTT broker and install the MQTT integration MQTT Broker
Copy .env.example to .env and replace the variable place-holders with your real configuration.
Make sure you switched the COMMAND_SOURCE in your .env file from "stdin" to "mqtt" and set the BROKER_* variables to connect to your MQTT broker address
Install docker compose plugin
Run docker compose up -d in the main folder of the application to run the ha-sip service

Now you can use the mqtt.publish service in home assistant to send commands as json to the hasip/execute topic from your automations

Example:

 service: mqtt.publish
 data:
     payload: >-
         { "command": "dial", "number": "sip:**[email protected]", "menu": { "message": "Hello from ha-sip.", "language": "en" } }
     topic: hasip/execute

You can listen to call state event on the topic configured in MQTT_STATE_TOPIC (defaults to hasip/state).

Support

If you find this project helpful, please consider giving it a star ⭐ on GitHub! Your support helps others discover the project and keeps me motivated.

Development

Create a virtual environment with pjsip and dependencies installed running ./build.sh create-venv from the root directory of the repo
Activate virtual env with source venv/bin/activate (bash, might be different with other shells)
Copy .env.example to .env and replace the variable place-holders with your real configuration.

HA_BASE_URL is something like "http://homeassistant.local:8123/api"

The access token is created from http://homeassistant.local:8123/profile
Run ./build.sh run-local to run the add-on locally

Paste commands as json (without line-breaks) into stdin of the running add-on:

Example:

{ "command": "dial", "number": "sip:**[email protected]", "menu": { "message": "Hello from ha-sip.", "language": "en" } }

Name		Name	Last commit message	Last commit date
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.github		.github
ha-sip		ha-sip
.editorconfig		.editorconfig
.env.example		.env.example
.gitignore		.gitignore
LICENSE		LICENSE
README.md		README.md
VOICE-ASSISTANT.md		VOICE-ASSISTANT.md
build.sh		build.sh
docker-compose.yaml		docker-compose.yaml
icon.png		icon.png
repository.json		repository.json

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ha-sip

Home Assistant SIP/VoIP Gateway is a Home Assistant add-on which

Installation

For global_options you can specify the following options

For options on each SIP account there are

Usage

Outgoing calls

To hang up the call again:

To send DTMF digits to an established call:

To transfer a call to a different SIP URI:

To bridge the audio streams of two active calls:

To play a message through TTS

To play an audio file

To stop audio playback (both audio file and message):

Incoming calls

Listen mode

Accept mode

Call menu definition

Web-hooks

incoming_call

call_established

entered_menu

dtmf_digit

call_disconnected

playback_done for message (TTS)

playback_done for audio file

ring_timeout

timeout

Examples

Trigger services through DTMF on an outgoing call

Play a message without DTMF interaction on sip account 1

Use PIN protection on outgoing call

Troubleshooting

Stand-alone mode

Support

Development

About

Resources

License

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