sdesalve
commented
8 months ago Closed f18m closed 8 months ago
sdesalve
commented
8 months ago 
f18m
commented
8 months ago hi @sdesalve , sorry but I still have some trouble. I looked at the other tickets you linked and I added the "tts" section in my configuration.yaml. Now I get the following error:
[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: 4.0.0
You are running the latest version of this add-on.
System: Debian GNU/Linux 12 (bookworm) (amd64 / qemux86-64)
Home Assistant Core: 2024.4.0
Home Assistant Supervisor: 2024.03.1
-----------------------------------------------------------
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] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:+393391300XXX@voip.eutelia.it","message_tts":"Prova messaggio"}
jq: error (at <stdin>:1): Cannot index number with string "url"
parse error: Invalid numeric literal at line 1, column 13
[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.do you know off-hand what that means?
f18m
commented
8 months ago I noticed that in the dss_psjua.log file the entry for the TTS service contains:
JSONGOOGLETTS = '500 Internal Server Error
Server got itself in trouble'I have no clue why it happens though
f18m
commented
8 months ago I did find in the HA logs the following stack trace associated with the 500 Internal Server Error:
Registratore: aiohttp.server
Fonte: /usr/local/lib/python3.12/site-packages/aiohttp/web_protocol.py:421
Prima occorrenza: 01:00:57 (1 occorrenze)
Ultima registrazione: 01:00:57
Error handling request
Traceback (most recent call last):
File "/usr/local/lib/python3.12/site-packages/aiohttp/web_protocol.py", line 452, in _handle_request
resp = await request_handler(request)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
File "/usr/local/lib/python3.12/site-packages/aiohttp/web_app.py", line 543, in _handle
resp = await handler(request)
^^^^^^^^^^^^^^^^^^^^^^
File "/usr/local/lib/python3.12/site-packages/aiohttp/web_middlewares.py", line 114, in impl
return await handler(request)
^^^^^^^^^^^^^^^^^^^^^^
File "/usr/src/homeassistant/homeassistant/components/http/security_filter.py", line 92, in security_filter_middleware
return await handler(request)
^^^^^^^^^^^^^^^^^^^^^^
File "/usr/src/homeassistant/homeassistant/components/http/forwarded.py", line 83, in forwarded_middleware
return await handler(request)
^^^^^^^^^^^^^^^^^^^^^^
File "/usr/src/homeassistant/homeassistant/components/http/request_context.py", line 26, in request_context_middleware
return await handler(request)
^^^^^^^^^^^^^^^^^^^^^^
File "/usr/src/homeassistant/homeassistant/components/http/ban.py", line 88, in ban_middleware
return await handler(request)
^^^^^^^^^^^^^^^^^^^^^^
File "/usr/src/homeassistant/homeassistant/components/http/auth.py", line 236, in auth_middleware
return await handler(request)
^^^^^^^^^^^^^^^^^^^^^^
File "/usr/src/homeassistant/homeassistant/components/http/headers.py", line 32, in headers_middleware
response = await handler(request)
^^^^^^^^^^^^^^^^^^^^^^
File "/usr/src/homeassistant/homeassistant/helpers/http.py", line 73, in handle
result = await handler(request, **request.match_info)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
File "/usr/src/homeassistant/homeassistant/components/tts/__init__.py", line 1017, in post
base = get_url(self.tts.hass)
^^^^^^^^^^^^^^^^^^^^^^
File "/usr/src/homeassistant/homeassistant/helpers/network.py", line 208, in get_url
raise NoURLAvailableError
homeassistant.helpers.network.NoURLAvailableErrorActually I realized the error was related to the wrong setting of the HA external URL. I have fixed that in my HA -> settings -> network panel. Next trouble is related to the call that is not going through... I get this:
[Info] Received messages {"call_sip_uri":"sip:3391300XXX@voip.eutelia.it","message_tts":"Prova messaggio"}
Converting audio file 'https://myurl.duckdns.org/api/tts_proxy/a772c970523ed7389b60b2ed1b674b3705d965e37_no_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:3391300XXX@voip.eutelia.it'...
This call will be terminated after '50' seconds.
01:05:48.530 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
01:05:48.530 sip_endpoint.c .Creating endpoint instance...
01:05:48.530 pjlib .select() I/O Queue created (0x7f76fc615100)
01:05:48.530 sip_endpoint.c .Module "mod-msg-print" registered
01:05:48.530 sip_transport.c .Transport manager created.
01:05:48.530 pjsua_core.c .PJSUA state changed: NULL --> CREATED
01:05:48.537 pjsua_core.c .pjsua version 2.11.1 for Linux-6.1.0.18/x86_64 initialized
01:05:48.539 pjsua_app.c .Turning sound device -99 -99 ON
01:05:48.539 main.c Ready: Success
01:05:48.540 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.4:5060>: does not register
Online status: Online
[ 1] <sip:172.30.33.4:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:0595964988@voip.eutelia.it: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:3391300XXX@voip.eutelia.it
+=============================================================================+
| 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:3391300XXX@voip.eutelia.it [CALLING]
>>> 01:05:49.539 pjsua_app.c .Turning sound device -99 -99 OFF
01:06:20.561 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 01:06:40.051 timer.c .Dumping timer heap:
01:06:40.051 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...getting closer..
sdesalve
commented
8 months ago Ok. Domanda stupida: c'è credito sull'account?
sdesalve
commented
8 months ago sip:3391300XXX@voip.eutelia.it
E dimmi che il numero scritto qui è corretto nella tua configurazione
f18m
commented
8 months ago Ok. Domanda stupida: c'è credito sull'account?
si c'e' credito, ho verificato. Sono anche riuscito ad usare il numero di telefono attivo tramite un softphone come Zoiper.
sip:3391300XXX@voip.eutelia.it E dimmi che il numero scritto qui è corretto nella tua configurazione
si si è corretto, l'ho solo oscurato per privacy su questa github issue...
Stasera posso tentare altre idee... ma attualmente sono un pò a corto di idee. Posso provare ad usare pjsua direttamente da command-line di un sistema Linux? Come posso lanciarlo per fare una chiamata base, avendo a disposizione un MP3 come audio?
grazie!!
sdesalve
commented
8 months ago metti
1 configurazione TTS 2 configurazione addon 3 log completo addon dall'avvio all'errore
f18m
commented
8 months ago metti
1 configurazione TTS 2 configurazione addon 3 log completo addon dall'avvio all'errore
ok certo, ecco la configurazione TTS:
tts:
- platform: google_translate
service_name: google_translate_say
language: "no"la configurazione dell'addon:
caller_id_uri: sip:0595964988@voip.eutelia.it
realm: "*"
username: "0595964988"
password: HIDDEN_JUST_IN_THIS_GITHUB_ISSUEe il log dell'addon:
[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: 4.0.0
You are running the latest version of this add-on.
System: Debian GNU/Linux 12 (bookworm) (amd64 / qemux86-64)
Home Assistant Core: 2024.4.0
Home Assistant Supervisor: 2024.03.1
-----------------------------------------------------------
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] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:3391300XXX@voip.eutelia.it","message_tts":"Prova messaggio"}
Converting audio file 'https://xxx.duckdns.org/api/tts_proxy/772c970523ed7389b60b2ed1b674b3705d965e37_no_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:3391300XXX@voip.eutelia.it'...
This call will be terminated after '50' seconds.
23:53:16.038 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
23:53:16.039 sip_endpoint.c .Creating endpoint instance...
23:53:16.039 pjlib .select() I/O Queue created (0x7f5c4893a100)
23:53:16.039 sip_endpoint.c .Module "mod-msg-print" registered
23:53:16.039 sip_transport.c .Transport manager created.
23:53:16.039 pjsua_core.c .PJSUA state changed: NULL --> CREATED
23:53:16.050 pjsua_core.c .pjsua version 2.11.1 for Linux-6.1.0.18/x86_64 initialized
23:53:16.060 pjsua_app.c .Turning sound device -99 -99 ON
23:53:16.061 main.c Ready: Success
23:53:16.070 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.4:5060>: does not register
Online status: Online
[ 1] <sip:172.30.33.4:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:0595964988@voip.eutelia.it: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:3391300XXX@voip.eutelia.it
+=============================================================================+
| 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:3391300XXX@voip.eutelia.it [CALLING]
>>> 23:53:17.061 pjsua_app.c .Turning sound device -99 -99 OFF
23:53:48.092 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 23:54:07.567 timer.c .Dumping timer heap:
23:54:07.567 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...Attenzione: mi e' capitato di leggere nel README che per certi operatori FreeVoipDeal/Any other Dellmont/Betamax serve come opzione '--no-tcp'. L'ho provata e ho ricevuto la chiamata!! finalmente!
Forse va aggiornata la docu per indicare che serve --no-tcp anche per Irideos/Eutelia ?
sdesalve
commented
8 months ago Fammi capire
language: "no"
Hai messo no per norvegese?
call_sip_uri":"sip:3391300XXX@voip.eutelia.it","message_tts":"Prova
Vuoi chiamare un +33 senza mettere il + o un +39 e lo stai omettendo?
Prova a mettere nelle pjsua options --no-tcp
f18m
commented
8 months ago Fammi capire
language: "no"
Hai messo no per norvegese?
Si mi sono sbagliato. Appena ho sentito l'audio ho realizzato e corretto in "it".
call_sip_uri":"sip:3391300XXX@voip.eutelia.it","message_tts":"Prova
Vuoi chiamare un +33 senza mettere il + o un +39 e lo stai omettendo?
un +39 e sto omettendo il prefisso. Dici che è meglio esplicitarlo?
Prova a mettere nelle pjsua options
--no-tcp
si con questa opzione è andata! Meglio aggiungerlo nel README?
Hi @sdesalve , I'm getting an error when I try to use the DSS VoIP Notifier. This is the complete log of the add-on:
(just note that I obscured the last 3 digits of the cellphone I'm calling for privacy reasons). Note that I'm using the service from the HA "Developer tools" -> Services tab: