[END OF LIFE - ABANDONED - DISMISSED] DSS VoIP Notifier

Hi Sdesalve,

First one, congratulations for your aportation... It will be a great advance in our alarm integrations.

I have tested it, but without good results.

I have the next error when i try to execute the script from the developers area.

I have tryed it on two different HA instances, one of both with google say working perfectly in spanish and in google home. The other one, is a new one instance, but with identical output.

![image|690x467](upload://661wuo47Hmfp9ZTLTxX53OMcVhd.png) 

What kind of error can i have??

thanks in advance…

First of all, Voipgain.com is a Betamax provider so you need to set option like stated in docs:
https://github.com/sdesalve/hassio-addons/tree/master/dss_voip#freevoipdealany-other-dellmontbetamax-provider-1

Secondly, obiuvsly you need to restart add-on after it’s config has been changed…

And last but not least you need to enable SIP calls from Voipgain setting.

Let me know if it works.

PS: in developers area, please try to use only text… Not placeholder like { nombre }

bye

Hi,

I have checked it...

My config is like the sample for betamax

i have restarted and rebuilded the add-on.

And you are right, i did not have enabled the SIP call, but i have enabled it now, and the problem persists.

The error sounds like a problem with the translation, or simialar.....

 I have tryed with url instead of translation of text, but shows the same error to me.

 Have you some idea of what can i make to continue=???

thanks in advance.

Please post your config (remove private data) and add-on’s logs…

HI sdesalve,

Now it's working.

I think the problem was solved when i enabled the sip calls.

It take about 30 minutes to work, and i don't wait enough.

Thanks for your aportation to the hassio community, it will be very very good for me and for others.
1 Like

Glad to have helped you!
I appreciate your words.

Bye

PS: donations/coffee are welcome! :grin::grin::grin:

Hi, very nice add-on.
I’m trying to make calls through my fritzbox instead of public voip provider. I configured hassio as an internal number. It registers correctly in my fritzbox and it makes calls. The problem is that I can’t hear any voice message (neither using tts service nor using local mp3). It seems that something is missing or it is not selecting a codec. In the fritz log I can see this:


Normally I see --> G711 / <-- G711
I also tried to force G711 codec by adding the custom pjsua option:
pjsua_custom_options: --add-codec=G711
Where I’m wrong? Can you please help me?
Thanks

Hello,

Try to disable SIP LAN filter in your Fritz box settings…

If you call other local SIP clients, you can ear anything?

Hi,
how did you configure the add-on. I will try it like you with my fritzbox.

Hi,
thank you for your reply.

This is my configuration:
Fritz
Model 7490; OS 7.12; standard internal number configuration (e.g. **625; username “homeassistant” and password “xxxxxxxxxx”; do not react to any external incoming call).
IP 192.168.178.1

Home Assistant
Version 0.103.5; Hassio on raspberry pi 3b+ with raspbian and docker.
IP 192.168.178.5

Dss Voip Notifier Add on
Version 3.1.0;
this is current the add on config:

{
“sip_parameters”: {
“caller_id_uri”: “sip:[email protected]”,
“sip_server_uri”: “sip:192.168.178.1”,
“realm”: “*”,
“username”: “homeassistant”,
“password”: “xxxxxxxxxx”
}
}

and these are the parameters used for calling the service hassio.addon_stdin:

{
“addon”: “89275b70_dss_voip”,
“input”: {
“call_sip_uri”: “sip:**[email protected]”,
“audio_file_url”: “url of the VoipMessage.mp3”,“max_call_time”:“20”
}
}

where **1 is the internal number of the analogic phone.

I receive a mute call both using local mp3 and tts service.
Opening the audio file url with a browser I can listen to the mp3 message.

Thanks for your help.

Hi,
thanks for your reply.
No I can’t but all other sip clients or voip devices work fine together.
What is SIP LAN filter? I don’t find that parameter in my fritzbox.
Thank you for your help.

There is an option to lock outgoing SIP packets from internal LAN devices. But it can be a problem if you want to use external providers.

Please have patience for some days. I’ve received today a Fritzbox 7490… I’ll try to configure it and I’ll give you my feedback.

Only thing that I can suggest you is to disable TCP transport, like a betamax provider

Thanks sdesalve,
I’ve already tried with --no-tcp option.
I suspect that is a problem of routing since I see that the service is calling from an IP different from my network (see tha image of the fritz log).


Could 172.30.33.2 be the virtual ip of the container of the add-on?
My local network is 192.168.178.0/24
Let me know about your test.
Bye

This option shoud be disabled: in other config with remote SIP server this option prevent outgoing SIP traffic

With my Fritz!Box 7490 I can call another local client and I ear TTS message.

This is my config:
192.168.178.136 = Raspberry IP
192.168.178.1 = Fritz!Box IP

Add-on’s Config:

{
  "sip_parameters": {
    "caller_id_uri": "sip:[email protected]:5060",
    "realm": "*",
    "username": "homeassistant",
    "password": "xxxx"
  },
  "pjsua_custom_options": "--no-tcp --ip-addr=192.168.178.136"
}

Service Call:

{
  "addon": "89275b70_dss_voip",
  "input": { "call_sip_uri":"sip:**[email protected]:5060","message_tts":"Prova messaggio" }
}

Let me know if works also with your setup.

bye

2 Likes

Great job!
Now it’s working perfectly.
The solution is this option:

  --ip-addr="ip of the raspberry pi"

Thanks a lot for your tip.

Finally, this is my working add-on’s configuration for fritzbox PBX:

{
  "sip_parameters": {
    "caller_id_uri": "sip:[email protected]",
    "sip_server_uri": "sip:192.168.178.1",
    "realm": "*",
    "username": "homeassistant",
    "password": "xxxxxxxxxxxx"
  },
  "pjsua_custom_options": "--ip-addr=192.168.178.5"
}

Where:
192.168.178.5 = Raspberry IP
192.168.178.1 = Fritz!Box IP

Thanks

2 Likes

This is crazy, it works very well.

Thanx for the addon!!

1 Like

Good idea! Beautiful work.
I can finally use it with my fritz! Box.

Congratulations!!! :phone:

1 Like

This is really cool! I need to if I can have it now calls through my FreeSwitch PBX software to the “paging” endpoints on the VoIP phones. Then I can play announcements through those phones in auto-answer speakerphone mode.

Try to configure it. Docs can help you.
You need only SIP Server address, username & password.

Like one of the other posters, I don’t run Hass.io, but Home Assistant in a docker container (along with a bunch of other containers). So I will need to cobble together something first…