chan_sip
was removed in Asterisk 21, thus it is no longer supported by this add-on. You should migrate tores_pjsip
if you were usingchan_sip
.
- Upgrade Asterisk from 20.8.1 to 22.1.0
- Upgrade debian-base from 7.4.0 to 7.6.2
- Add custom startup script support
- Prefer Opus over other codecs (by @OnFreund at #331)
- Upgrade Asterisk from 20.6.0 to 20.8.1
- Upgrade debian-base from 7.3.3 to 7.3.4
- Use
friendly_name
instead ofid
as caller id (by @OnFreund at #322) - Fix
/config/asterisk/custom
files being overwritten on container restart (by @felipecrs at #323, fixes #309)- PS: this bug did not affect people running the add-on with Home Assistant Supervisor, only for people running the add-on as a standalone docker container.
- Include
app_rtsp_sip
in ARM builds
- Upgrade Asterisk from 20.5.2 to 20.6.0
- Upgrade asterisk-chan-dongle to latest revision
- Upgrade asterisk-googletts to latest revision
- Upgrade Asterisk from 20.4.0 to 20.5.2
- Upgrade hassio-addons/debian-base from 7.0.1 to 7.1.0
- Fix missing
libunbound8
library forres_resolver_unbound
-
Add
res_resolver_unbound
module, which allows to change the DNS server for Asterisk. Example:; /config/asterisk/custom/resolver_unbound.conf [general] nameserver = 127.0.0.1 resolv =
- Fix Asterisk Mailbox Server missing
lame
- Ensure Asterisk Mailbox Server is started with
--verbose
if log level is debug or higher - Fix Asterisk Mailbox Server media directory to
/media/asterisk/voicemail/default/<mailbox-extension>/
for persistency - Fix Asterisk Mailbox Server cache file to
/data/tmp
for persistency - Fix to the
/data
directory (and also/media
when needed) are created on startup, useful when running the add-on as a standalone docker container - Fix the permissions of downloaded sounds to be
0755
to be as indicated by https://asterisksounds.org, although on my tests it was working before as well.
- Upgrade hassio-addons/debian-base from 6.2.7 to 7.0.1
- This upgrades Debian 11 to Debian 12
- Upgrade Asterisk from 20.2.1 to 20.4.0
- Update asterisk_mbox_server
- Added IPv6 support
- Patch upstream asterisk issue
- Upgrade Asterisk from 20.2.0 to 20.2.1
- Upgrade debian-base from 6.2.3 to 6.2.5
- Fix add-on not starting for ARM users
- Fix
auto_add_secret
validation always failing as if it was net set
Some default options for the add-on configuration have been switched:
generate_ssl_cert
is now enabled by default.video_support
is now disabled by default. It barely worked anyway.
Make sure to check the add-on configuration page after updating the add-on to ensure your configuration is still correct.
Now the add-on can be run as a standalone docker container:
docker pull ghcr.io/tech7fox/asterisk-hass-addon:4.0.0
- An example
docker-compose.yml
file is available here. - Make sure to mount a
config
folder to/config
and amedia
folder to/media
to ensure the add-on can access your configuration and media files. - To configure the add-on options you can use the
/config/config.json
file. The default options can be seen here. - If you enable
auto_add
to automatically create extensions for every Person in your Home Assistant, make sure to also set:- The
HA_TOKEN
environment variable with your Home Assistant long-lived access token - The
HA_URL
, unless https://homeassistant.local:8123 resolves to your Home Assistant instance
- The
Also, you can now use absolute paths in the certfile
and keyfile
options.
The add-on no longer sends discovery information for the Asterisk integration in Home Assistant. This never worked anyway, and if in the future it does, we can restore it.
There is now a new option for the add-on: Additional Sounds Languages to Download.
With this option, you can specify a list of languages to download sounds from https://www.asterisksounds.org/ on the add-on startup.
These sounds will be downloaded to /media/asterisk
, and the add-on will not download the sounds in case they were downloaded already.
These sounds will be available to use in Asterisk by changing the language as you would do normally. For example, you can put pt-BR
in the list of additional sounds to download, and then change the Asterisk configuration to use pt_BR
as language.
Finally, now the add-on is able to access files on /media
, which means you can store your custom music and sounds there, and then refer to them in the Asterisk configuration files.
All the available options will now appear in the add-on configuration page without having to click in Show unused options, which was an actually misleading name.
Also, the default log level is now INFO instead of NOTICE, which increases the logging a little bit.
-
The add-on now supports the
hassio.stdin
Home Assistant service to execute any Asterisk CLI commands. For example, to reload changes from/config/asterisk/custom/extensions.conf
:service: hassio.addon_stdin data: addon: b35499aa_asterisk input: dialplan reload
This means that you can now use the full power of Asterisk CLI right from your Home Assistant automations!
-
Use symbolic links to map custom Asterisk config files
- Previously, the custom Asterisk config files would be copied over the default files on container startup
- With the new approach, for example, the Asterisk CLI command to reload extensions after changing the
/config/asterisk/custom/extensions.conf
will work without requiring to restart the whole add-on.
- Fix
asterisk_mbox.ini
configuration again
- Fix
asterisk_mbox.ini
configuration
We changed the way we handle the Asterisk config files and this will require a manual action on your side. Now, Asterisk files you intend to modify should be placed under /config/asterisk/custom
. For example, if you were previously editing extensions.conf
, you should move it from /config/asterisk/extensions.conf
to /config/asterisk/custom/extensions.conf
.
After moving all the files you need to /config/asterisk/custom
, you can also cleanup the /config/asterisk
folder by deleting everything under it, except for the custom
folder.
Previously, both the default and custom Asterisk config files were being written and read from /config/asterisk
, which posed some issues related to upgrading the add-on as the config files written by an old version of the add-on would get read by the new version of the add-on as if they were user customized files. This also meant that, if users wanted to receive the new Asterisk default config files, they would have to delete everything from /config/asterisk
that was not customized manually before starting the container.
This is no longer required, and now default files will be always upgraded, while still retaining custom files between upgrades. This will require a manual action from you if you are upgrading this add-on from previous versions, see above.
- The default Asterisk config files are now copied to
/config/asterisk/default
on every container start. The files on this folder should be used for reference only, as any changes made in this folder will be overwritten in the container startup. - The custom Asterisk config files are now read from
/config/asterisk/custom
instead of from/config/asterisk
. - You can now override/customize any Asterisk files (previously, the auto-generated Asterisk files could not be overriden).
- Bump Asterisk from 20.1.0 to 20.2.0
- Bump debian-base from 6.2.0 to 6.2.3
- Add
chan_sip
(disabled by default) for Dahua VTO compatibility (by @bdherouville)
Delete the old sip.conf
and modules.conf
. This disables chan_sip
by default and sets it on another port to prevent conflicts with pjsip
.
- Upgrade Asterisk from 20.0.1 to 20.1.0 (by @felipecrs)
- Fix tmp dir for googletts and speech-recog (by @felipecrs)
- Now they use
/data/tmp
instead oftmp
, which is retained between restarts (but deleted upon uninstall) for add-ons.
- Now they use
- Refactor the installation of all patches for easier maintenance (by @felipecrs)
- Upgrade Asterisk from 18.15.0 to 20.0.1 (by @felipecrs)
- Upgrade addon-debian-base from 6.1.3 to 6.2.0 (by @felipecrs)
- Fix add-on failing to start sometimes (by @felipecrs)
- Upgrade Asterisk from 18.14.0 to 18.15.0 (by @felipecrs)
- Upgrade addon-debian-base from 6.1.1 to 6.1.3 (by @felipecrs)
- Upgrade asterisk-chan-dongle to 503dba8 (by @felipecrs)
- Upgrade Asterisk from 18.12.1 to 18.14.0 (by @felipecrs)
- Upgrade addon-debian-base from 6.0.0 to 6.1.1 (by @felipecrs)
- Only include rtsp-sip for amd64 and i386 (by @TECH7Fox)
- Add php (by @TECH7Fox)
- Add rtsp-sip (by @TECH7Fox)
- Update builder and linter (by @felipecrs)
- Add pt-BR translations (by @LeandroIssa)
- Upgrade addon-base from 5.3.0 to 6.0.0 (by @felipecrs)
- Upgrade Asterisk from 18.10.1 to 18.12.1 (by @felipecrs)
- Fix domain bug that makes the WS contacts unreachable (by @TECH7Fox)
- Increase maximum possible number of SDP formats (#140) (by @nanosonde)
- Fix TLS transport
- Set NAT settings
- Include default STUN server
- Fix timezone
- Disable qualify for the generated pjsip extensions, because it wasn't used and caused problems
- Do not load chan-dongle by default (because it seems to cause lots of errors and warnings when there is no dongle attached)
- You have to delete the
/config/asterisk/modules.conf
file so that the new one which has disabled chan-dongle can be created.
- You have to delete the
- Add asterisk-chan-dongle
- Add asterisk-googletts
- Add asterisk-speech-recog
- Run Asterisk as root (instead of as asterisk), this is requires so that chan-dongle can properly communicate with the dongle
- Tidy up some minor things
More information at #124.
- Fix Asterisk never starting after starting the addon (again, but for a different reason this time)
- Fix Asterisk never starting after starting the addon, which started to happen after v2.0.0 (issue #127, pr #128)
- Change base from Alpine to Debian (#116) (by @nanosonde)
- Addon size has been considerable increased
- Upgrade Asterisk to 18.1.0 (#116) (by @nanosonde)
- Now we build it from source, so we can always use the latest version and have more control about it
- Migrate from
chan_sip
tores_pjsip
(#112) (by @nanosonde)- This is a breaking change. Check below the upgrade guide.
Lots of issues were fixed by the above.
It's strongly recommended to erase your existing Asterisk configuration before upgrading.
- Move any customization you have done in
/config/asterisk/
to somewhere else. - Delete the
/config/asterisk
folder. - Restore your customizations to the
/config/asterisk
folder if you have any. - Make sure to convert your extensions from
chan_sip
tores_pjsip
if you have any.
Then, you can go ahead and upgrade. Next time you start the addon, it will recreate the files at /config/asterisk
.
- Include hint settings and add busylevel to auto generated extensions.
- Fix verbose and debug log levels
- Add
log_level
option
- Remove the initial Ingress support added in 1.2.0.
- Ingress will not be needed to make the integration and the card work without having to export additional ports or configuring additional reverse proxies (details here).
- Remove the option to disable SSL (#98)
- Disabling SSL causes the HA-SIP card not to work anymore.
- Add an option to automatically generate a self-signed certificate (#98)
- So that, users running Asterisk behind a reverse proxy do not need to bother about managing their own certificate to be used by Asterisk. You can directly proxy
https://<ha-ip>:8089
.
- So that, users running Asterisk behind a reverse proxy do not need to bother about managing their own certificate to be used by Asterisk. You can directly proxy
- Disable WS protocol wrongly introduced in 1.2.0 which caused issues
- Add an option to disable SSL (#66)
- Useful for setting up Asterisk to work behind a reverse proxy like NGINX. Here is one example on how to configure NGINX to proxy the Asterisk WebSockets interface: https://warlord0blog.wordpress.com/2020/04/16/asterisk-webrtc/
- PS: This was not tested, so any feedback is welcome.
- Add initial support for Ingress (#57)
- Work is still required from other components like the integration and the card to effectively be able to use it.
- Note that the WebUI shown in the addon page is not a GUI page, but rather the WebSocket connection needed by SIP.JS to connect to the SIP server.
- Fix mailbox server not working (#92)
- Disable docker builtin init, to prevent multiple init systems as we already have S6 Overlay (#89)
- Fix permission denied error in discovery service (#85)
- Send out discovery information.
- Add
host
andport
to discovery.
- Optimize when mailbox service is disabled #80
- Fix translations
- Add mailbox server #68. To use with the Asterisk Mailbox Integration.
- Add discovery for the Asterisk Integration.
- Allow to customize
logging.conf
.
- Use parking instead of conference.
- Remove default passwords.
- Update config and docs.
- Add default conference room #54. Now you can join a conference room via 444 (for default user) or 555 (for admin user). This is useful for things like doorbells.
- Add music-on-hold. (moh)
- Add video support #39. This feature comes disabled by default as otherwise the SIP Lovelace Card does not work in the companion app.
- Fix AMI permit. Now use
localhost
as host in the Asterisk integration. - Add auto_add_secret option #50. This option is to prevent having a default secret for the auto-added extensions.
- Allow custom configs using
/config/asterisk/sip_custom.conf
- Fix
_displayName
errors from SIP Lovelace Card
- Use prebuilt images for faster installation
- Use S6 Overlay to manage the Asterisk service
Check the commit history.