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| Skip to content Navigation Menu Toggle navigation Sign in Appearance settings * Platform + GitHub Copilot Write better code with AI + GitHub Spark New Build and deploy intelligent apps + GitHub Models New Manage and compare prompts + GitHub Advanced Security Find and fix vulnerabilities + Actions Automate any workflow + Codespaces Instant dev environments + Issues Plan and track work + Code Review Manage code changes + Discussions Collaborate outside of code + Code Search Find more, search less Explore + Why GitHub + Documentation + GitHub Skills + Blog Integrations + GitHub Marketplace + MCP Registry View all features * Solutions By company size + Enterprises + Small and medium teams + Startups + Nonprofits By use case + App Modernization + DevSecOps + DevOps + CI/CD + View all use cases By industry + Healthcare + Financial services + Manufacturing + Government + View all industries View all solutions * Resources Topics + AI + DevOps + Security + Software Development + View all Explore + Learning Pathways + Events & Webinars + Ebooks & Whitepapers + Customer Stories + Partners + Executive Insights * Open Source + GitHub Sponsors Fund open source developers + The ReadME Project GitHub community articles Repositories + Topics + Trending + Collections * Enterprise + Enterprise platform AI-powered developer platform Available add-ons + GitHub Advanced Security Enterprise-grade security features + Copilot for business Enterprise-grade AI features + Premium Support Enterprise-grade 24/7 support * Pricing Search or jump to... Search code, repositories, users, issues, pull requests... Search Clear Search syntax tips Provide feedback We read every piece of feedback, and take your input very seriously. Include my email address so I can be contacted Cancel Submit feedback Saved searches Use saved searches to filter your results more quickly Name Query To see all available qualifiers, see our documentation. Cancel Create saved search Sign in Sign up Appearance settings Resetting focus You signed in with another tab or window. Reload to refresh your session. You signed out in another tab or window. Reload to refresh your session. You switched accounts on another tab or window. Reload to refresh your session. Dismiss alert {{ message }} arnonym / ha-plugins Public * Uh oh! There was an error while loading. Please reload this page. * Notifications You must be signed in to change notification settings * Fork 31 * Star 268 Home-Assistant SIP Gateway License Apache-2.0 license 268 stars 31 forks Branches Tags Activity Star Notifications You must be signed in to change notification settings * Code * Issues 14 * Pull requests 1 * Actions * Projects 0 * Security Uh oh! There was an error while loading. Please reload this page. * Insights Additional navigation options * Code * Issues * Pull requests * Actions * Projects * Security * Insights arnonym/ha-plugins main Branches Tags Go to file Code Open more actions menu Folders and files Name Name Last commit message Last commit date Latest commit History 100 Commits .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 View all files Repository files navigation * README * Apache-2.0 license 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:homeassistant@fritz.box 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:anotheruser@fritz.box 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:**620@fritz.box # number to call. Format depends on your SIP provider, # but might look like 'sip:+49123456789@fritz.box' 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:**620@fritz.box To send DTMF digits to an established call: service: hassio.addon_stdin data: addon: c7744bff_ha-sip input: command: send_dtmf number: sip:**620@fritz.box 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:**620@fritz.box transfer_to: sip:**623@fritz.box To bridge the audio streams of two active calls: service: hassio.addon_stdin data: addon: c7744bff_ha-sip input: command: bridge_audio number: sip:**620@fritz.box bridge_to: sip:**623@fritz.box To play a message through TTS service: hassio.addon_stdin data: addon: c7744bff_ha-sip input: command: play_message number: sip:**620@fritz.box 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:**620@fritz.box 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:**620@fritz.box 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:5551234456@fritz.box> " , "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:5551234456@fritz.box> " , "parsed_caller": " 5551234456 " , "sip_account": 1, "internal_id": " something-unique " } call_established { "event": " call_established " , "caller": " <sip:5551234456@fritz.box> " , "parsed_caller": " 5551234456 " , "sip_account": 1, "internal_id": " something-unique " } entered_menu { "event": " entered_menu " , "caller": " <sip:5551234456@fritz.box> " , "parsed_caller": " 5551234456 " , "menu_id": " owner " , "sip_account": 1, "internal_id": " something-unique " } dtmf_digit { "event": " dtmf_digit " , "caller": " <sip:5551234456@fritz.box> " , "parsed_caller": " 5551234456 " , "digit": " 1 " , "sip_account": 1, "internal_id": " something-unique " } call_disconnected { "event": " call_disconnected " , "caller": " <sip:5551234456@fritz.box> " , "parsed_caller": " 5551234456 " , "sip_account": 1, "internal_id": " something-unique " } playback_done for message (TTS) { "event": " playback_done " , "caller": " <sip:5551234456@fritz.box> " , "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:5551234456@fritz.box> " , "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:5551234456@fritz.box> " , "parsed_caller": " 5551234456 " , "sip_account": 1, "internal_id": " something-unique " } timeout { "event": " timeout " , "caller": " <sip:5551234456@fritz.box> " , "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:**620@fritz.box 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:**620@fritz.box 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:**620@fritz.box 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. 1. Follow the instructions from home assistant to set up a working MQTT broker and install the MQTT integration MQTT Broker 2. Copy .env.example to .env and replace the variable place-holders with your real configuration. 3. 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 4. Install docker compose plugin 5. Run docker compose up -d in the main folder of the application to run the ha-sip service 6. 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:**620@fritz.box", "menu": { "message": "Hello from ha-sip.", "language": "en" } } topic: hasip/execute 7. 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 1. Create a virtual environment with pjsip and dependencies installed running ./build.sh create-venv from the root directory of the repo 2. Activate virtual env with source venv/bin/activate (bash, might be different with other shells) 3. 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 4. Run ./build.sh run-local to run the add-on locally 5. Paste commands as json (without line-breaks) into stdin of the running add-on: Example: { "command": " dial " , "number": " sip:**620@fritz.box " , "menu": { "message": " Hello from ha-sip. " , "language": " en " } } About Home-Assistant SIP Gateway Topics automation sip home-assistant Resources Readme License Apache-2.0 license Uh oh! There was an error while loading. Please reload this page. Activity Stars 268 stars Watchers 10 watching Forks 31 forks Report repository Sponsor this project Sponsor Uh oh! There was an error while loading. Please reload this page. Learn more about GitHub Sponsors Uh oh! There was an error while loading. Please reload this page. Contributors 4 * arnonym Arne Gellhaus * Hypercookie Jannes Müller * TheFes Martijn van der Pol * tizzen33 Languages * Python 90.8% * Shell 8.3% * Dockerfile 0.9% Footer © 2025 GitHub, Inc. Footer navigation * Terms * Privacy * Security * Status * Community * Docs * Contact * Manage cookies * Do not share my personal information You can’t perform that action at this time. | Skip to content Navigation Menu Toggle navigation Sign in Appearance settings * Platform + GitHub Copilot Write better code with AI + GitHub Spark New Build and deploy intelligent apps + GitHub Models New Manage and compare prompts + GitHub Advanced Security Find and fix vulnerabilities + Actions Automate any workflow + Codespaces Instant dev environments + Issues Plan and track work + Code Review Manage code changes + Discussions Collaborate outside of code + Code Search Find more, search less Explore + Why GitHub + Documentation + GitHub Skills + Blog Integrations + GitHub Marketplace + MCP Registry View all features * Solutions By company size + Enterprises + Small and medium teams + Startups + Nonprofits By use case + App Modernization + DevSecOps + DevOps + CI/CD + View all use cases By industry + Healthcare + Financial services + Manufacturing + Government + View all industries View all solutions * Resources Topics + AI + DevOps + Security + Software Development + View all Explore + Learning Pathways + Events & Webinars + Ebooks & Whitepapers + Customer Stories + Partners + Executive Insights * Open Source + GitHub Sponsors Fund open source developers + The ReadME Project GitHub community articles Repositories + Topics + Trending + Collections * Enterprise + Enterprise platform AI-powered developer platform Available add-ons + GitHub Advanced Security Enterprise-grade security features + Copilot for business Enterprise-grade AI features + Premium Support Enterprise-grade 24/7 support * Pricing Search or jump to... Search code, repositories, users, issues, pull requests... Search Clear Search syntax tips Provide feedback We read every piece of feedback, and take your input very seriously. Include my email address so I can be contacted Cancel Submit feedback Saved searches Use saved searches to filter your results more quickly Name Query To see all available qualifiers, see our documentation. Cancel Create saved search Sign in Sign up Appearance settings Resetting focus You signed in with another tab or window. Reload to refresh your session. You signed out in another tab or window. Reload to refresh your session. You switched accounts on another tab or window. Reload to refresh your session. Dismiss alert {{ message }} arnonym / ha-plugins Public * Uh oh! There was an error while loading. Please reload this page. * Notifications You must be signed in to change notification settings * Fork 31 * Star 268 Home-Assistant SIP Gateway License Apache-2.0 license 268 stars 31 forks Branches Tags Activity Star Notifications You must be signed in to change notification settings * Code * Issues 14 * Pull requests 1 * Actions * Projects 0 * Security Uh oh! There was an error while loading. Please reload this page. * Insights Additional navigation options * Code * Issues * Pull requests * Actions * Projects * Security * Insights arnonym/ha-plugins main Branches Tags Go to file Code Open more actions menu Folders and files Name Name Last commit message Last commit date Latest commit History 100 Commits .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 View all files Repository files navigation * README * Apache-2.0 license 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:homeassistant@fritz.box 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:anotheruser@fritz.box 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:**620@fritz.box # number to call. Format depends on your SIP provider, # but might look like 'sip:+49123456789@fritz.box' 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:**620@fritz.box To send DTMF digits to an established call: service: hassio.addon_stdin data: addon: c7744bff_ha-sip input: command: send_dtmf number: sip:**620@fritz.box 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:**620@fritz.box transfer_to: sip:**623@fritz.box To bridge the audio streams of two active calls: service: hassio.addon_stdin data: addon: c7744bff_ha-sip input: command: bridge_audio number: sip:**620@fritz.box bridge_to: sip:**623@fritz.box To play a message through TTS service: hassio.addon_stdin data: addon: c7744bff_ha-sip input: command: play_message number: sip:**620@fritz.box 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:**620@fritz.box 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:**620@fritz.box 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:5551234456@fritz.box> " , "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:5551234456@fritz.box> " , "parsed_caller": " 5551234456 " , "sip_account": 1, "internal_id": " something-unique " } call_established { "event": " call_established " , "caller": " <sip:5551234456@fritz.box> " , "parsed_caller": " 5551234456 " , "sip_account": 1, "internal_id": " something-unique " } entered_menu { "event": " entered_menu " , "caller": " <sip:5551234456@fritz.box> " , "parsed_caller": " 5551234456 " , "menu_id": " owner " , "sip_account": 1, "internal_id": " something-unique " } dtmf_digit { "event": " dtmf_digit " , "caller": " <sip:5551234456@fritz.box> " , "parsed_caller": " 5551234456 " , "digit": " 1 " , "sip_account": 1, "internal_id": " something-unique " } call_disconnected { "event": " call_disconnected " , "caller": " <sip:5551234456@fritz.box> " , "parsed_caller": " 5551234456 " , "sip_account": 1, "internal_id": " something-unique " } playback_done for message (TTS) { "event": " playback_done " , "caller": " <sip:5551234456@fritz.box> " , "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:5551234456@fritz.box> " , "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:5551234456@fritz.box> " , "parsed_caller": " 5551234456 " , "sip_account": 1, "internal_id": " something-unique " } timeout { "event": " timeout " , "caller": " <sip:5551234456@fritz.box> " , "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:**620@fritz.box 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:**620@fritz.box 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:**620@fritz.box 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. 1. Follow the instructions from home assistant to set up a working MQTT broker and install the MQTT integration MQTT Broker 2. Copy .env.example to .env and replace the variable place-holders with your real configuration. 3. 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 4. Install docker compose plugin 5. Run docker compose up -d in the main folder of the application to run the ha-sip service 6. 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:**620@fritz.box", "menu": { "message": "Hello from ha-sip.", "language": "en" } } topic: hasip/execute 7. 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 1. Create a virtual environment with pjsip and dependencies installed running ./build.sh create-venv from the root directory of the repo 2. Activate virtual env with source venv/bin/activate (bash, might be different with other shells) 3. 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 4. Run ./build.sh run-local to run the add-on locally 5. Paste commands as json (without line-breaks) into stdin of the running add-on: Example: { "command": " dial " , "number": " sip:**620@fritz.box " , "menu": { "message": " Hello from ha-sip. " , "language": " en " } } About Home-Assistant SIP Gateway Topics automation sip home-assistant Resources Readme License Apache-2.0 license Uh oh! There was an error while loading. Please reload this page. Activity Stars 268 stars Watchers 10 watching Forks 31 forks Report repository Sponsor this project Sponsor Uh oh! There was an error while loading. Please reload this page. Learn more about GitHub Sponsors Uh oh! There was an error while loading. Please reload this page. Contributors 4 * * * * Uh oh! There was an error while loading. Please reload this page. Languages * Python 90.8% * Shell 8.3% * Dockerfile 0.9% Footer © 2025 GitHub, Inc. Footer navigation * Terms * Privacy * Security * Status * Community * Docs * Contact * Manage cookies * Do not share my personal information You can’t perform that action at this time. |
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