for some reason during addon install your docker cannot resolve DNS for alpinelinux.org packages repository.
Please try to change DNS on your router to 8.8.8.8, reboot your router and also your HA host machine
disable also firewall/ad remover to try
i have hidden the numbers by XX, log says audio is created successfully, tts service works ok for me… do i have to add base_url ?
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...
error is [reason=403 (Forbidden)]… Is your login details correct?
also try with this
tried with outbound again, nothing…
login is correct, username = sipnumber password = sipnumber
@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?
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
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
no, on my Raspberry and on my miniPC without a Soundcard it runs
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
How do you access your hassio GUI?
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
Have you resolved your issue?
Feedback will be appreciated… And also post your providers configuration for sharing it on add-on’s docs
Have you resolved your issue?
Feedback will be appreciated… And also post your providers configuration for sharing it on add-on’s docs
No, sorry, I gave up
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
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.
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.
config/www not /config/local !!! 