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

Share your Projects!
, , ,
23

Ok, I think did all as per instructions but receive this error


PJSUA_CUSTOM_OPTIONS = '--outbound=sip:sip.messagenet.it:5061;lr'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri": "sip:[email protected]", "message_tts": "Attenzione e' scattato l'allarme in via xxxxx"}
Converting audio file 'https://xxx.duckdns.org:8123/api/tts_proxy/xxxx5d3b51a423b0b2e7282xxxxxx_it_-_google_translate.mp3'...
[cmd] /run.sh exited 2
[cont-finish.d] executing container finish scripts...
[cont-finish.d] 99-message.sh: executing... 
-----------------------------------------------------------
                Oops! Something went wrong.

 We are so sorry, but something went terribly wrong when
 starting or running this add-on.
 
 Be sure to check the log above, line by line, for hints.
-----------------------------------------------------------
[cont-finish.d] 99-message.sh: exited 0.
[cont-finish.d] done.
[s6-finish] waiting for services.
[s6-finish] sending all processes the TERM signal.

this is the config and the scripot


{
  "sip_parameters": {
    "caller_id_uri": "sip:[email protected]",
    "realm": "*",
    "username": "53xxx",
    "password": "xxx",
    "sip_server_uri": "sip:sip.messagenet.it:5061"
  },
  "pjsua_custom_options": "--outbound=sip:sip.messagenet.it:5061;lr"
}


  allarme_voip_kla:
    sequence:
    - service: hassio.addon_stdin
      data_template:
        addon: 89275b70_dss_voip
        input: {"call_sip_uri":"sip:[email protected]","message_tts":"Attenzione e' scattato l'allarme in via xxx"}
24

Ahh I think I know

this command


https://xxx.duckdns.org:8123/api/tts_proxy/xxxx5d3b51a423b0b2e7282xxxxxx_it_-_google_translate.mp3

should be in my case


https://xxx.duckdns.org/api/tts_proxy/xxxx5d3b51a423b0b2e7282xxxxxx_it_-_google_translate.mp3

(without the port number). Any way I can fix this?

26

As my Pvt message you should arrange base_url key in your Google TTS settings

See

1 Like
27

Yes! That’s how it should be


tts:
  - platform: google_translate
    service_name: google_say
    cache: true
    cache_dir: /tmp/tts
    time_memory: 300
    language: 'it'
    base_url: https://xxx.duckdns.org
1 Like
28

A new article in Italian language with a tutorial for Italian user that helps to install and configure this add-on:

DSS VoIP Notifier https://hassiohelp.eu/2019/08/28/ricevere-chiamate-da-homeassistant-dss-voip-notifier/

Add phone calls to your Hassio setup, Have a look!

2 Likes
29

This looks amazing!! I would 100% have a use for this, but I cant get it installed:

19-08-28 09:49:24 INFO (SyncWorker_9) [hassio.docker.addon] Start build 89275b70/amd64-addon-dss_voip:1.1.0
19-08-28 09:49:38 ERROR (SyncWorker_9) [hassio.docker.addon] Can't build 89275b70/amd64-addon-dss_voip:1.1.0: The command '/bin/ash -o pipefail -c set -o pipefail         && echo '@edge http://dl-cdn.alpinelinux.org/alpine/edge/main' >> /etc/apk/repositories     && echo '@edge http://dl-cdn.alpinelinux.org/alpine/edge/community' >> /etc/apk/repositories     && echo '@edge http://dl-cdn.alpinelinux.org/alpine/edge/testing' >> /etc/apk/repositories         && apk add --no-cache --virtual .build-dependencies         tar=1.32-r0         && apk add --no-cache         libcrypto1.1=1.1.1c-r0         libssl1.1=1.1.1c-r0         musl-utils=1.1.22-r2         musl=1.1.22-r2         && apk add --no-cache         bash=5.0.0-r0         curl=7.65.1-r0         jq=1.6-r0         tzdata=2019a-r0         && apk add --no-cache         sox=14.4.2-r5         pjsua=2.8-r0         screen=4.6.2-r0         && S6_ARCH="${BUILD_ARCH}"     && if [ "${BUILD_ARCH}" = "i386" ]; then S6_ARCH="x86"; fi     && if [ "${BUILD_ARCH}" = "armv7" ]; then S6_ARCH="arm"; fi         && curl -L -s "https://github.com/just-containers/s6-overlay/releases/download/v1.22.1.0/s6-overlay-${S6_ARCH}.tar.gz"         | tar zxvf - -C /         && mkdir -p /etc/fix-attrs.d     && mkdir -p /etc/services.d         && curl -J -L -o /tmp/bashio.tar.gz         "https://github.com/hassio-addons/bashio/archive/v0.3.2.tar.gz"     && mkdir /tmp/bashio     && tar zxvf         /tmp/bashio.tar.gz         --strip 1 -C /tmp/bashio         && mv /tmp/bashio/lib /usr/lib/bashio     && ln -s /usr/lib/bashio/bashio /usr/bin/bashio         && apk del --purge .build-dependencies     && rm -f -r         /tmp/*' returned a non-zero code: 1

Any help why it wont build? :frowning:

Would be happy to share my automations when I finish getting it up and running!

(I have plans to build a ‘get out of situation’ bot that can call on request and playback a message such as family emergency or something!)

30

What kind of system are you using?
I’ve tested it on ResinOS/HassOS on Raspberry 3b.
Other users have installed it on Debian+Docker+Hassio and Ubuntu+Docker+Hassio on Intel NUC without problems…

Should be compatible with everything, but let me know other details about your setup…

31

Hello, i’m trying to install the addon but i receive this error.
I’m running Hassio through VirtualBox

19-08-28 12:51:14 INFO (SyncWorker_3) [hassio.docker.addon] Start build 89275b70/amd64-addon-dss_voip:1.1.0
19-08-28 12:51:33 ERROR (SyncWorker_3) [hassio.docker.addon] Can't build 89275b70/amd64-addon-dss_voip:1.1.0: The command '/bin/ash -o pipefail -c set -o pipefail         && echo '@edge http://dl-cdn.alpinelinux.org/alpine/edge/main' >> /etc/apk/repositories     && echo '@edge http://dl-cdn.alpinelinux.org/alpine/edge/community' >> /etc/apk/repositories     && echo '@edge http://dl-cdn.alpinelinux.org/alpine/edge/testing' >> /etc/apk/repositories         && apk add --no-cache --virtual .build-dependencies         tar=1.32-r0         && apk add --no-cache         libcrypto1.1=1.1.1c-r0         libssl1.1=1.1.1c-r0         musl-utils=1.1.22-r2         musl=1.1.22-r2         && apk add --no-cache         bash=5.0.0-r0         curl=7.65.1-r0         jq=1.6-r0         tzdata=2019a-r0         && apk add --no-cache         sox=14.4.2-r5         pjsua=2.8-r0         screen=4.6.2-r0         && S6_ARCH="${BUILD_ARCH}"     && if [ "${BUILD_ARCH}" = "i386" ]; then S6_ARCH="x86"; fi     && if [ "${BUILD_ARCH}" = "armv7" ]; then S6_ARCH="arm"; fi         && curl -L -s "https://github.com/just-containers/s6-overlay/releases/download/v1.22.1.0/s6-overlay-${S6_ARCH}.tar.gz"         | tar zxvf - -C /         && mkdir -p /etc/fix-attrs.d     && mkdir -p /etc/services.d         && curl -J -L -o /tmp/bashio.tar.gz         "https://github.com/hassio-addons/bashio/archive/v0.3.2.tar.gz"     && mkdir /tmp/bashio     && tar zxvf         /tmp/bashio.tar.gz         --strip 1 -C /tmp/bashio         && mv /tmp/bashio/lib /usr/lib/bashio     && ln -s /usr/lib/bashio/bashio /usr/bin/bashio         && apk del --purge .build-dependencies     && rm -f -r         /tmp/*' returned a non-zero code: 1

Can you help me?
Thanks a lot
Stefano

32

I see that you are using an AMD64 system… Let me investigate why setup fail!
I’ll get in touch with you asap

33

Someone here who has an own sip server running ? Don’t wanna use public one

34

Thanks !

Yep, Hass.io running on Ubuntu 18.04.3 on an i5 NUC!

35

I have a 3CX server running that I will be trying to set it up with !

36

Do you have a link for me ? Would like setting it up also

37

Really? https://www.3cx.com/

1 Like
38

I’ve released a fix. Please upgrade and let me know if now it install…

bye

39 
{
  "sip_parameters": {
    "caller_id_uri": "sip:[email protected]",
    "realm": "*",
    "username": "AuthenticationID",
    "password": "AuthenticationPassword"
  }
}

working on latest update!!

Heres my 3CX settings.

Can I have it call non sip numbers? an outbound call to a regular number?

Also, how hard would it be to simply pass a .mp3 file through to be played ? or use a different TTS provider (I want to use google wavenet!)

edit:

You can break it by calling the wrong number… lol

Audio succesfully converted...
Starting SIP Client and calling 'tel:<myphonenumber>'...
12:33:06.524         os_core_unix.c !pjlib 2.8 for POSIX initialized
12:33:06.524         sip_endpoint.c  .Creating endpoint instance...
12:33:06.525                  pjlib  .select() I/O Queue created (0x558cd809ca70)
12:33:06.525         sip_endpoint.c  .Module "mod-msg-print" registered
12:33:06.525        sip_transport.c  .Transport manager created.
12:33:06.525           pjsua_core.c  .PJSUA state changed: NULL --> CREATED
12:33:06.533           pjsua_core.c  .pjsua version 2.8 for Linux-4.15.0.58/x86_64 initialized
12:33:06.534            pjsua_app.c  Error adding buddy: Invalid URI scheme (PJSIP_EINVALIDSCHEME)
./run: line 212: echo: write error: Broken pipe
[cont-finish.d] executing container finish scripts...
[cont-finish.d] 99-message.sh: executing... 
[cont-finish.d] 99-message.sh: exited 0.
[cont-finish.d] done.
[s6-finish] waiting for services.
[s6-finish] sending all processes the TERM signal.

edit2:

passing a PSTN number works as follows:

  {
      "addon": "89275b70_dss_voip",
      "input": {
        "call_sip_uri": "sip:[email protected]",
        "message_tts": "This is a test call, please disregard"
      }
    }
1 Like
40

Cannot seem to get this to work with Linphone so far. I think it just comes down to the fact that i’m not especially experienced when it comes to sip/voip configuration. So if anyone gets it to work with Linphone in the future, i would gladly appreciate any posts with configs!

41

Please post your config and add-on’s log

42

Here is my config (replaced with username instead of my own ofc)

{
  "sip_parameters": {
    "caller_id_uri": "sip:[email protected]",
    "realm": "*",
    "username": "username",
    "password": "password",
    "sip_server_uri": "sip.linphone.org"
  }
}

And here is the startup log

 Hass.io Add-on: DSS VoIP Notifier
 VoIP Notifier for HomeAssistant
-----------------------------------------------------------
 Add-on version: 2.0.1
 You are running the latest version of this add-on.
 System: HassOS 1.13  (amd64 / qemux86-64)
 Home Assistant version: 0.98.1
 Supervisor version: 184
-----------------------------------------------------------
 Please, share the above information when looking for help
 or support in, e.g., GitHub, forums or the Discord chat.
-----------------------------------------------------------
[cont-init.d] 00-banner.sh: exited 0.
[cont-init.d] 01-log-level.sh: executing... 
[cont-init.d] 01-log-level.sh: exited 0.
[cont-init.d] done.
[services.d] starting services
[services.d] done.
[Info] Starting addon...
[Info] Registering as SIP Client...
-----------------------------------------------------------
                  SIP Client registered.

 Call <sip:[email protected]>/VoIP phone number
 to check system status.
 You'll find logs in /share/dss_voip/dss_pjsua_autoanswer.log
-----------------------------------------------------------
[Info] Listening for messages via stdin service call...

So it seems to register but when i make this service call

  - service: hassio.addon_stdin
    data_template:
      addon: 89275b70_dss_voip
      input: {"call_sip_uri":"sip:[email protected]","message_tts":"Welcome, the door is unlocked"}

This happens:

16:40:42.244         os_core_unix.c !pjlib 2.8 for POSIX initialized
16:40:42.244         sip_endpoint.c  .Creating endpoint instance...
16:40:42.245                  pjlib  .select() I/O Queue created (0x5646d0209a70)
16:40:42.245         sip_endpoint.c  .Module "mod-msg-print" registered
16:40:42.245        sip_transport.c  .Transport manager created.
16:40:42.245           pjsua_core.c  .PJSUA state changed: NULL --> CREATED
16:40:42.309           pjsua_core.c  .pjsua version 2.8 for Linux-4.14.82/x86_64 initialized
16:40:42.310            pjsua_app.c  .Turning sound device -99 -99 ON
16:40:42.310                 main.c  Ready: Success
16:40:42.324            pjsua_app.c  .......Call 0 state changed to CALLING
>>>>
Account list:
  [ 0] <sip:172.30.33.6:5060>: does not register
       Online status: Online
  [ 1] <sip:172.30.33.6:5060;transport=TCP>: does not register
       Online status: Online
 *[ 2] sip:[email protected]: does not register
       Online status: Online
Buddy list:
 [ 1] <?>  sip:[email protected]
+=============================================================================+
|       Call Commands:         |   Buddy, IM & Presence:  |     Account:      |
|                              |                          |                   |
|  m  Make new call            | +b  Add new buddy       .| +a  Add new accnt |
|  M  Make multiple calls      | -b  Delete buddy         | -a  Delete accnt. |
|  a  Answer call              |  i  Send IM              | !a  Modify accnt. |
|  h  Hangup call  (ha=all)    |  s  Subscribe presence   | rr  (Re-)register |
|  H  Hold call                |  u  Unsubscribe presence | ru  Unregister    |
|  v  re-inVite (release hold) |  t  ToGgle Online status |  >  Cycle next ac.|
|  U  send UPDATE              |  T  Set online status    |  <  Cycle prev ac.|
| ],[ Select next/prev call    +--------------------------+-------------------+
|  x  Xfer call                |      Media Commands:     |  Status & Config: |
|  X  Xfer with Replaces       |                          |                   |
|  #  Send RFC 2833 DTMF       | cl  List ports           |  d  Dump status   |
|  *  Send DTMF with INFO      | cc  Connect port         | dd  Dump detailed |
| dq  Dump curr. call quality  | cd  Disconnect port      | dc  Dump config   |
|                              |  V  Adjust audio Volume  |  f  Save config   |
|  S  Send arbitrary REQUEST   | Cp  Codec priorities     |                   |
+-----------------------------------------------------------------------------+
|  q  QUIT   L  ReLoad   sleep MS   echo [0|1|txt]     n: detect NAT type     |
+=============================================================================+
You have 1 active call
Current call id=0 to sip:[email protected] [CALLING]
>>> 16:40:42.356            pjsua_app.c  .....Call 0 is DISCONNECTED [reason=407 (Proxy Authentication Required)]
16:40:42.356     pjsua_app_common.c  .....
  [DISCONNCTD] To: sip:[email protected]
    Call time: 00h:00m:00s, 1st res in 46 ms, conn in 0ms
16:40:43.311            pjsua_app.c  .Turning sound device -99 -99 OFF
[Info] Call ended...
[Info] Listening for messages via stdin service call...

It does not connect to my intercom, seems to be a proxy error right?

43

SIP Client should register correctly. I need to see dss_pjsua_autoanswer.log file in your /share/dss_voip folder.
Try to call [email protected] to test if it has been registered

As you stated error is related with proxy auth
pjsua_app.c …Call 0 is DISCONNECTED [reason=407 (Proxy Authentication Required)]

I’m searching PjSua option to activate it, if possible