[New Addon] DSS VoIP Notifier: Hassio makes phone calls to you! ☎️

1 I will assume that you have a Fritz.box router with VoIP functionality. Your LAN IP CLASS is 192.168.178.xx

2 you don’t need to specify sip_server_uri options. You can safely remove it from addon config

3 in your configuration.yaml or in you automations.yaml you need to invoke this addon to place a call. So you’ll have something like:

service

so i have a Script but when save come error:

Message malformed: Integration not found.

:frowning:

please read HomeAssistant documentation and try to make a simple automation before try to use more complex thing.

I don’t use GUI script generator so I cannot help you with this:

WOW its working, thanks

alias: Anruf
sequence:

  • service: hassio.addon_stdin
    data:
    addon: 89275b70_dss_voip34
    input:
    call_sip_uri: ‘sip:---------------@fritz.box
    message_tts: Alarm

So what is the best way to call five peaple…my Solution…(at the moment with two)

alias: Anruf call
description: ‘’
trigger:

  • platform: state
    entity_id: light.homematik_dimmaktor_fur_markenschalter
    from: ‘0’
    to: ‘1’
    condition: []
    action:
  • service: script.1615833818761(Anruf)
    mode: single

alias: Anruf Autom.
sequence:

  • service: hassio.addon_stdin
    data:
    addon: 89275b70_dss_voip34
    input:
    call_sip_uri: ‘sip:number [email protected]
    message_tts: Alarm
  • wait_template: ‘’
    timeout: ‘00:01:00’
  • service: hassio.addon_stdin
    data:
    addon: 89275b70_dss_voip34
    input:
    call_sip_uri: ‘sip:number [email protected]
    message_tts: Alarm
    mode: queued
    max: 3

thanks

1 Like

You can make 5 service call… Or you can use a Virtual pbx like pbxes.com to make an extension that will make ringing all sip client you like…

Hi,
I have successfully made the call. However, the error log is still recorded in the log and the call reception time is quite slow, I don’t know the cause. I appreciate any help (I use GRANDSTREAM UCM6202)

You have 1 active call
Current call id=0 to sip:[email protected] [CALLING]
>>> 11:46:09.457            pjsua_app.c  .Turning sound device -99 -99 OFF
**11:46:39.919     tcpc0x7fe4a9052508  TCP connect() error: [code=120110]: Operation timed out**
**11:46:39.919      tsx0x7fe4a90686b8  Temporary failure in sending Request msg INVITE/cseq=2804 (tdta0x7fe4a9061a88), will try next server: Operation timed out**
**11:46:39.919            pjsua_app.c  SIP TCP transport is disconnected from xxx.xxx.xxx.xxx:5060: Operation timed out [status=120110]**
11:46:40.241            pjsua_app.c  ........Turning sound device -99 -99 ON
11:46:40.241            pjsua_app.c  .....Call 0 state changed to EARLY (180 Ringing)
11:46:43.184            pjsua_app.c  .....Call 0 state changed to CONNECTING
11:46:43.185            pjsua_app.c  .....Call 0 state changed to CONFIRMED
11:46:45.296            pjsua_app.c  ......Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
11:46:45.296     pjsua_app_common.c  ......
  [DISCONNCTD] To: sip:[email protected];tag=12ad81cb-4556-4dd2-9aa0-80f1a4b7c3bd
    Call time: 00h:00m:02s, 1st res in 31785 ms, conn in 34729ms
    #0 audio PCMU @8kHz, sendrecv, peer=xxx.xxx.xxx.xxx:18952
       SRTP status: Not active Crypto-suite: 
       RX pt=0, last update:00h:00m:01.838s ago
          total 103pkt 16.4KB (20.6KB +IP hdr) @avg=63.5Kbps/79.4Kbps
          pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
                (msec)    min     avg     max     last    dev
          loss period:   0.000   0.000   0.000   0.000   0.000
          jitter     :   0.000   0.709   1.750   1.250   0.362
       TX pt=0, ptime=20, last update:never
          total 90pkt 14.4KB (18.0KB +IP hdr) @avg=55.5Kbps/69.4Kbps
          pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
                (msec)    min     avg     max     last    dev 
          loss period:   0.000   0.000   0.000   0.000   0.000
          jitter     :   0.000   0.000   0.000   0.000   0.000
       RTT msec      :   0.000   0.000   0.000   0.000   0.000

What does this mean? “quite slow” is not a technical term.

1 Like

and there isn’t any info about your system config… Router, IP Class, addon startup log ecc

try, I see a TCP error

pjsua_custom_options: '--no-tcp'
1 Like

Has anyone managed to use this in a plain docker install, no hass OS or hassio or whatever, just home-assistant in docker.

I built the image but it is looking for supervisor etc. I was just hoping to spin up a docker container in the usual way.

Add-ons uses Supervisor to get commands and give output. Basically it’s a bash script that call an executable.

You can edit the “run” bash script to obtain parameters from command line and place the desired call…

i used this a lot when i had google translate, but now i use nabu casa tts and i need to set the language and gender in my calls- but how to do this in this addon?. i dont have any config in configuration.yaml when i use nabu casa. i set the tts in the addon config and it works but i need to set the gender and language.

Set nabucasa tts as you need and then

The thing is that I want the sip voice to be English and male. But for other automations I use danish and female.

You can Play an MP3 audio file…
If you configure your hassio tts to synthetize in danish female voice you can’t get tts in other languages…

Or you can use another tts platform configured in different way…

add

platform: cloud

to use NABU CASA TTS Service

Thank you very much sdesalve for your really handy addon and all the hard work, you’ve put into creating this. It’s working perfectly with my Fritzbox and is currently used in Fire Alert-Automations, as well as in a self created “Panic Mode”.
Is there any way in showing the CLIP-number to the callers, when i am calling someone through the addon? The number seems to be always supressed, even though i set up the FritzBox to generally showing the numbers for outbound calls. When using the SIP-Number with other calling tools, the number will be shown. Is it a configuration thing in the addon itself, or is it the wrong place to look into in finding it out?
Thanks in advance and stay safe.
Chris

Would you try to set

caller_id_uri: 'sip:[email protected]'

With your fritzbox line phone number?

eg:

caller_id_uri: 'sip:+393331234567'

try also without country code or without leading +

Let me know if this solve your issue

Thanks for the fast reply.
I just tried out all of the constellations, but of no luck. When used with a trailing plus the addon refuses to start at all based on the following error: “pjsua_app_config.c Error: invalid SIP URL ‘sip:+49301234567’ in local id argument”
All the other number variants were used by the addon and with a test call through the dev tools according to the protocol of the addon it was in the state of “CALLING”, but on fritzbox-side there were no sip-registrations active, or any call attempts visible.

try

caller_id_uri: 'sip:[email protected]'

ID is the value that pjsip will use for FROM: header…

This time it accepted the plus in the id, but nevertheless no calls went through to my phone.
I again tried out all the ways possible how to write my number, but no combination worked in the end.
Thanks for your help so far sdesalve. I really appreciate your support.

I actually would be intrigued to know, if i’m the only one here with a surpressed caller id (in combination with a fritzbox).