Hello, any help or idee, how i can intecrate the access modul from unifi on my dashboard? What could be a right configuration for the SIP?
I have successfully installed it, I am trying to use it for BAS-IP to be able to make calls from assistant to my Intercom and vice versa, these are my configurations:
and I have this automation with an action to be able to make an outgoing call:
from the addon logs, everything seems fine:
but I am not getting a call to the number I am trying to call, I have a sip client installed on my laptop but didn’t receive a call. What am I missing?
Hi. Is this addon abandoned? There hasn’t been any activity from the codeowner on github since the end of last year so I am wondering…
It’s not abandoned, but because of lacking time I wasn’t able to implement new features. And it’s working for me, so there’s that.
Thank you very much for confirming that it’s still alive. I’m using it myself since 2 months and I am amazed at how easy it makes it to generate phone calls with home assistant. It’s an amazing thing. I only have the odd issue that approximately 1 out of 10 calls doesn’t work due to a segmentation error in the addon. I looked in the issues on github and saw it already mentioned but noticed that there was no progress in solving the issue, hence why I asked.
I looked into that many times already and I could not figure out a solution so far.
I’m not a good enough programer myself to be of any help for this. Maybe a solution or workaround will present itself at a later point in time. Thanks again for your great work.
I finally got it working on docker with the mqtt command source.
I struggled for days since I was not using the “next” branch.
Now I can make calls and play TTS audio files. Since I’m using piper I had to use chime_tts integration with the “chime_tts.say_url” to get the audio file to play in the call.
I’d like to have a topic with the call status too, to know if the call has been answered or has been hang up.
Hi Fabio72,
I created a new beta version, which includes a new topic to send state events to when using the MQTT broker connection. Please test and write an issue when something’s not working as expected.
Arne
I actually saw this message like three quarters of a year too late, but this should be possible with bridge_audio. Currently not sure how to wait for the second person to pick up, but it might be no problem to call this function for each person, and it just won’t bridge for the first call.
No worries, I found a solution
https://github.com/TheFes/HA-configuration/blob/84e216b9330bd8eb187995f7c315c4131880bb21/packages/00_ground_floor/living_room/phone_menu.yaml#L122
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I tried to recreate the image but I’m getting strange errors on bashio/log.sh and the container stops.
I tried to restore the previous version (I saved the folder and I recreated the image) but now I’m getting the same error.
I’m unable to get it working again 
nothing else changed. No updates of OS or home assistant container.
I was able to create the container again, with the Debian image.
But I’m facing the same problem I had when I first started to play with it:
The container is up, says “Running standalone version of ha-sip”, but it doesn’t register to my asteriskr nor to the broker.
Some people on github told me, the current version runs much more stable for them. Maybe you want to give it a try!
I got a workaround to make it work again.
The problem seems related to .env file non getting interpretated correctly.
Removing all quotes from .env file makes the pbx registration successfull.
In order to have the mqtt part working too, I had to patch the mqtt.py file, hardcoding all variables and removing all “os.environ.get” calls.
Now is 100% working, with the state topic too.
Hallo, i use the plugin in germany. I like to use my sip number to listen for incomming call. My provider is the german telekom. Withe evry setting i get the fail
sip_auth_client.c ...Unable to set auth for tdta0x1d05ea28: can not find credential for tel.t-online.de/Digest MD5
pjsua_acc.c ....SIP registration error: No suitable credential (PJSIP_ENOCREDENTIAL) [status=171101]
OnRegState: 401 Unauthorized
can somebody help me ?
@arnonym
I’m trying to call a phone using the sip addon, and then play an audio file.
I first tried the dial command, then bridge it to the HA number and then play the audio. That doesn’t work.
In idle mode, when I pick up the phone, there is a menu, if that is active, the play_audio_file command works, so I know that one is correct.
In the logs I noticed this
| 12:59:33.216711 [ ] Got "dial" command for sip:**[email protected]
| &id001
| id: null
| message: null
| audio_file: null <=====
| language: en
| action: null
| choices_are_pin: false
| choices: {}
| default_choice:
| id: null
| message: Unknown option
| audio_file: null
| language: en
| action: null
| choices_are_pin: false
| choices: null
| default_choice: null
| timeout_choice: null
This seems to imply I can immediately use an audio file in dial action. However, this also doesn’t work:
- alias: Call old phone
action: hassio.addon_stdin
data:
addon: c7744bff_ha-sip
input:
command: dial
number: "sip:**[email protected]"
audio_file: /config/www/SinterklaasTelefoon.mp3
cache_audio: true
wait_for_audio_to_finish: true
webhook_to_call:
What’s the best way to do this?
Hmm, this seems to be similar as Need help: Call -> Play mp3 -> TTS -> Play mp3 · Issue #153 · arnonym/ha-plugins · GitHub
I now have this:
script:
piet_call:
alias: Piet Call
sequence:
- alias: Call old phone
action: hassio.addon_stdin
data:
addon: c7744bff_ha-sip
input:
command: dial
number: "sip:**[email protected]"
audio_file: /config/www/SinterklaasTelefoon.mp3
cache_audio: true
wait_for_audio_to_finish: true
webhook_to_call:
playback_done: piet_mp3_finished
call_established: piet_call_established
- alias: "Wait for call connection"
wait_for_trigger:
- alias: "Call established webhook"
trigger: webhook
local_only: true
webhook_id: piet_call_established
- alias: short delay
delay: 0.5
- alias: Play mp3
action: hassio.addon_stdin
data:
addon: c7744bff_ha-sip
input:
command: play_audio_file
number: "**621"
audio_file: /config/www/SinterklaasTelefoon.mp3
cache_audio: true
wait_for_audio_to_finish: true
- alias: "Wait for sound to finish"
wait_for_trigger:
- alias: "Call established webhook"
trigger: webhook
local_only: true
webhook_id: piet_mp3_finished
- alias: "Hang up the remaining caller"
action: hassio.addon_stdin
data:
addon: c7744bff_ha-sip
input:
command: hangup
number: "sip:**[email protected]"
solved it already, for some reason it worked with only **621 in devtools > actions, but in the script I had to use sip:**[email protected]
Can you use the add-on to call to HA for to use Assist?
The number to call is *47192*168*1*100*5060?
I have a phone menu so the kids can call us parents, and the grandparents, but I’d like to add an option for Assist voice commands