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

Luisico · January 28, 2021, 5:37pm

Nothing, same issue.
I have added base_url to tts in configuration.yaml

Log here:

[Info] Starting addon...
PJSUA_CUSTOM_OPTIONS = '--no-tcp --proxy=sip:10.31.255.134:5070;lr'
[Info] Registering as SIP Client...
PJSUA_CUSTOM_OPTIONS = '--no-tcp --proxy=sip:10.31.255.134:5070;lr'
-----------------------------------------------------------
                  SIP Client registered.

 Call <sip:[email protected]>/VoIP phone number
 to check system status.
 You'll find logs in /share/dss_voip/dss_autoanswer.log
-----------------------------------------------------------
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri": "sip:[email protected]", "message_tts": "Write here your message"}
Converting audio file 'http://192.168.1.18:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_es_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '50' seconds.
18:34:12.941         os_core_unix.c !pjlib 2.9 for POSIX initialized
18:34:12.941         sip_endpoint.c  .Creating endpoint instance...
18:34:12.942                  pjlib  .select() I/O Queue created (0x7fccbf2530f0)
18:34:12.942         sip_endpoint.c  .Module "mod-msg-print" registered
18:34:12.942        sip_transport.c  .Transport manager created.
18:34:12.942           pjsua_core.c  .PJSUA state changed: NULL --> CREATED
18:34:12.962           pjsua_core.c  .pjsua version 2.9 for Linux-4.19.0.13/x86_64 initialized
18:34:12.965            pjsua_app.c  .Turning sound device -99 -99 ON
18:34:12.965                 main.c  Ready: Success
18:34:12.966            pjsua_app.c  .......Call 0 state changed to CALLING
>>>>
Account list:
  [ 0] <sip:172.30.33.7:5060>: does not register
       Online status: Online
 *[ 1] 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]
>>> 18:34:13.002            pjsua_app.c  .....Call 0 is DISCONNECTED [reason=403 (Forbidden)]
18:34:13.002     pjsua_app_common.c  .....
  [DISCONNCTD] To: sip:[email protected]
    Call time: 00h:00m:00s, 1st res in 37 ms, conn in 0ms
18:34:13.966            pjsua_app.c  .Turning sound device -99 -99 OFF
No current call
[Info] Call ended...
[Info] Listening for messages via stdin service call...

sdesalve · January 28, 2021, 5:40pm

error is [reason=403 (Forbidden)]… Is your login details correct?

sdesalve · January 28, 2021, 5:42pm

also try with this

Luisico · January 28, 2021, 5:45pm

tried with outbound again, nothing…

login is correct, username = sipnumber password = sipnumber

sdesalve · January 28, 2021, 5:51pm

@luisico have you placed a call to your VoIP number? Is addon answering to Your calls?
Also note that you can use VoIP login credential on 1 device at same time…

So you have a password that is equal to your phone number? True???

On /share/dss_voip/dss_pjsua.log is there any other info?

sdesalve · January 28, 2021, 6:04pm

sip:[email protected]:5070

try to add :5070

and also try att to '/etc/hosts’ on your Debian GNU/Linux 10 (buster) (amd64 / qemux86-64)

entry
213.4.130.95 telefonica.net

Luisico · January 28, 2021, 6:10pm

probably because no sound card installed in my debian… (i didnt tought to use it)

tried # sudo apt-get install alsa-utils

ALSA lib confmisc.c:767:(parse_card) cannot find card '0'
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory
ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1246:(snd_func_refer) error evaluating name
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5233:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM sysdefault
ALSA lib confmisc.c:767:(parse_card) cannot find card '0'
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory
ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1246:(snd_func_refer) error evaluating name
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5233:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM sysdefault
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.front
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround21
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround21
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround40
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround41
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround50
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround51
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.surround71
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.iec958
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.iec958
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.iec958
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.modem
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.phoneline
ALSA lib confmisc.c:767:(parse_card) cannot find card '0'
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory
ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1246:(snd_func_refer) error evaluating name
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5233:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM default
ALSA lib confmisc.c:767:(parse_card) cannot find card '0'
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory
ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1246:(snd_func_refer) error evaluating name
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5233:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM default
ALSA lib confmisc.c:767:(parse_card) cannot find card '0'
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_card_driver returned error: No such file or directory
ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_concat returned error: No such file or directory
ALSA lib confmisc.c:1246:(snd_func_refer) error evaluating name
ALSA lib conf.c:4745:(_snd_config_evaluate) function snd_func_refer returned error: No such file or directory
ALSA lib conf.c:5233:(snd_config_expand) Evaluate error: No such file or directory
ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM dmix
Cannot connect to server socket err = No such file or directory
Cannot connect to server request channel
jack server is not running or cannot be started
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock
JackShmReadWritePtr::~JackShmReadWritePtr - Init not done for -1, skipping unlock

sdesalve · January 28, 2021, 6:12pm

no, on my Raspberry and on my miniPC without a Soundcard it runs

Roto_Dos · January 30, 2021, 2:02pm

I have only one problem with the addon.
It fails to access the file generated by the tts.google_translate_say service or any audio file hosted in /local and transform it. I can play perfectly the audio file in the browser, and the files generated by google_tts exists and are accessible.
This is my call to the service with a audio file:

addon: 89275b70_dss_voip
input: {"call_sip_uri":"sip:[email protected]:5060","audio_file_url":"https:/XXX.duckdns.org/local/alarm.wav"}

And this is the output:

[s6-init] making user provided files available at /var/run/s6/etc...exited 0.
[s6-init] ensuring user provided files have correct perms...exited 0.
[fix-attrs.d] applying ownership & permissions fixes...
[fix-attrs.d] done.
[cont-init.d] executing container initialization scripts...
[cont-init.d] 00-banner.sh: executing... 
-----------------------------------------------------------
 Add-on: DSS VoIP Notifier
 VoIP Notifier for Home Assistant
-----------------------------------------------------------
 Add-on version: 3.5.5
 You are running the latest version of this add-on.
 System: Debian GNU/Linux 10 (buster)  (amd64 / qemux86-64)
 Home Assistant Core: 2021.1.5
 Home Assistant Supervisor: 2021.01.7
-----------------------------------------------------------
 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...
PJSUA_CUSTOM_OPTIONS = '--no-tcp'
[Info] Registering as SIP Client...
PJSUA_CUSTOM_OPTIONS = '--no-tcp'
-----------------------------------------------------------
                  SIP Client registered.

 Call <sip:[email protected]>/VoIP phone number
 to check system status.
 You'll find logs in /share/dss_voip/dss_autoanswer.log
-----------------------------------------------------------
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri": "sip:[email protected]:5060", "audio_file_url": "https://XXX.duckdns.org/local/alarm.wav"}
Converting audio file 'https://XXX.duckdns.org/local/alarm.wav'...

I tried another mp3 file hosted in another server and everything goes succesfull. Also tried from the addon console to wget the mp3 or a wav file hosted in home assistant /local and this is the output:

root@89275b70-dss-voip:/$ wget https://xxxx.duckdns.org/local/alarm.wav
--2021-01-30 14:56:40--  https://xxxx.duckdns.org/local/alarm.wav
Resolving xxxx.duckdns.org (xxxx.duckdns.org)... 109.xx.xx.xx
Connecting to xxxx.duckdns.org (xxxx.duckdns.org)|109.xx.xx.xx|:443... failed: Operation timed out.
Retrying.

If I make a wget from a mp3 hosted in another server also goes perfect.
Any idea?
Thank you

sdesalve · January 30, 2021, 2:41pm

How do you access your hassio GUI?

https://XXX.duckdns.org/

Perhaps Do you have exposed 8123 port and so you’ll need to specify port on your audio file URL?
https://XXX.duckdns.org:8123/local/alarm.wav

I think you’ll solve your problems with correct URL and specifying a base_url in your Google tts configuration

sdesalve · January 30, 2021, 2:46pm

Have you resolved your issue?

Feedback will be appreciated… And also post your providers configuration for sharing it on add-on’s docs

sdesalve · January 30, 2021, 2:47pm

Have you resolved your issue?

Feedback will be appreciated… And also post your providers configuration for sharing it on add-on’s docs

Luisico · January 30, 2021, 2:59pm

No, sorry, I gave up

Roto_Dos · January 30, 2021, 3:01pm

I use ngnix proxy manager and I only expose external tcp/udp 443 to internal ip of home assistant and internal port 8123. My access to GUI is https://xxx.duckdns.org/.

Mi tts configuration in configuration.yaml is:

tts:
  - platform: google_translate
    service_name: google_translate_say
    language: 'es'
    base_url: https://XXX.duckdns.org

jpenyc · January 30, 2021, 4:11pm

Just wanted to drop a line to say thanks again to the developer and all those who have contributed to this. I have had it set up in the house for a couple of months now and it is working like a charm.
I got some Grandstream phones real cheap on eBay, stuck them Poe where avail, and WiFi where not in all the rooms and now have a nice little voice for HA to tell me all kinds of useless information when I don’t need it.
I plan to integrate dial in commands to trigger automations etc. The phones have touchscreens, but are stuck on android 4 or some older version cant seem to do much with. For what I paid for them (<$10 ea), they are well worth it for what they are currently able to do.
I have a couple of newer Grandstreams with 7in touchscreens that i can run the HA android app on with no issues.
For the Pbx side of things, I’m running a dedicated remote host running the Incredible PBX system.

derwolff · January 30, 2021, 4:33pm

No sorry I can’t resolve the issue. I get the same error message:

The requested versions are not provided for my installation. I have no idea how to fix it.

sdesalve · January 30, 2021, 5:00pm

config/www not /config/local !!!

sdesalve · January 30, 2021, 5:02pm

have you tried to change local network DNS? It’s HassOS that fail to reach Alpinelinux repositories

sdesalve · January 30, 2021, 5:09pm

Nice feature to have in a pbx/home automation system!
Unfortunately PjSua can’t recognize and manage dial tone…
But I think can be developed with PjSIP libs and AppDaemon app… But is not in my plans for next future…

Thanks for your appreciation

jpenyc · January 30, 2021, 5:27pm

It would be doing something different to integrate than using this add-on. The PBX dial plan would connect directly to HA via scripts. (That’s how i remember doing it the last time a few years ago last time i ran my own Pbx and was using Indigo home auto setup).
Something like: dial 731 to turn a group of lights on, 732 to turn them off etc, sends appropriate triggers to HA