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sdesalve / hassio-addons

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DSS Notifier with google tts and fritzbox 7530 #91

Closed Ciqsky closed 6 months ago

Ciqsky commented 6 months ago

When I receive a phone call from my home number (I use a Fritzbox 7530 and a Raspberry Pi4) the TTS message is not always reproduced well. Let me explain better: if upon receiving the call on my phone number I answer immediately then the voice of the TTS message does not is played. If instead of responding immediately I wait 4/5 seconds it often works and the message is played. Have you already encountered this type of problem? Can it be resolved in some way? Thanks

sdesalve commented 6 months ago

Do you use this?

--ip-addr=RASPBERRY_IP_ADDRESS

Ciqsky commented 6 months ago

this my conf

caller_id_uri: sip:homeassistant@192.168.1.1:5060 realm: "*" username: xxxxxx password: xxxxxxx sip_server_uri: sip:192.168.1.1:5060 --no-tcp --ip-addr=192.168.1.105

this is my script example

service: hassio.addon_stdin data_template: input: call_sip_uri: sip:+xxxxxxxxxx@192.168.1.1:5060 message_tts: Non posso attivare allarme, problema con un sensore data: addon: xxxxxxx_dss_voip

sdesalve commented 6 months ago

Logs

And screenshot of addon config... It's strange what you have writed

Ciqsky commented 6 months ago

sip_parameters: caller_id_uri: sip:homeassistant@192.168.1.1:5060 realm: "*" username: xxx password: xxx sip_server_uri: sip:192.168.1.1:5060 pjsua_custom_options: "--no-tcp --ip-addr=192.168.1.105"

addon config

Ciqsky commented 6 months ago

You have 1 active call Current call id=0 to sip:xxxx@192.168.1.1:5060 [CALLING]

17:00:18.640 tsx0x7faef956d8 .......Temporary failure in sending Request msg INVITE/cseq=24205 (tdta0x7faef8eaa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT) 17:00:18.669 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) 17:01:08.386 pjsua_app_common.c ... [EARLY] To: sip:xxxx@192.168.1.1;tag=9EF2DAAB1C6321B2 Call time: 00h:00m:00s, 1st res in 35 ms, conn in 0ms

0 audio G722 @16kHz, sendrecv, peer=192.168.1.1:7078

SRTP status: Not active Crypto-suite: RX pt=9, last update:00h:00m:00.001s ago total 2.0Kpkt 321.9KB (402.4KB +IP hdr) @avg=51.8Kbps/64.7Kbps 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 0.023 0.625 0.000 0.062 TX pt=9, ptime=20, last update:00h:00m:12.829s ago total 1.9Kpkt 309.7KB (387.2KB +IP hdr) @avg=49.8Kbps/62.3Kbps 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 : 1.375 2.484 2.875 2.875 0.610 RTT msec : 0.717 0.805 0.946 0.778 0.086 17:01:08.387 pjsua_app.c .Call 0 is DISCONNECTED [reason=487 (Request Terminated)] 17:01:08.904 sip_transaction.c .....Unable to register CANCEL transaction (key exists) 17:01:08.904 pjsua_call.c ...Failed to send end session message: Object already exists (PJ_EEXISTS) [status=70015] 17:01:09.386 pjsua_app.c ..Turning sound device -99 -99 OFF 17:01:09.945 timer.c .Dumping timer heap: 17:01:09.945 timer.c . Cur size: 1 entries, max: 3070 17:01:09.945 timer.c . Entries: 17:01:09.945 timer.c . _id Id Elapsed Source 17:01:09.945 timer.c . ---------------------------------- 17:01:09.945 timer.c . 4 1 3.959 ../src/pjsua-lib/pjsua_call.c:2848 [Info] Call ended... [Info] Listening for messages via stdin service call...

sdesalve commented 6 months ago

Full addon logs

Altrimenti non vedo se sta rilevando le opzioni

sdesalve commented 6 months ago

E comunque metti uno screenshot censurato della configurazione... Magari non l'hai messa bene

Ciqsky commented 6 months ago

[Info] Received messages {"call_sip_uri":"sip:+39xxx@192.168.1.1:5060","message_tts":"Non posso attivare allarme, problema con un sensore"} Converting audio file 'http://192.168.1.105:8123/api/tts_proxy/0f1fcd1ded38411701c4934fe3b5527fdb3debb2_it_-_google_translate.mp3'... Audio succesfully converted... Starting SIP Client and calling 'sip:+39xxx@192.168.1.1:5060'... This call will be terminated after '50' seconds. 17:00:18.542 os_core_unix.c !pjlib 2.11.1 for POSIX initialized 17:00:18.544 sip_endpoint.c .Creating endpoint instance... 17:00:18.545 pjlib .select() I/O Queue created (0x7fafcfc100) 17:00:18.545 sip_endpoint.c .Module "mod-msg-print" registered 17:00:18.545 sip_transport.c .Transport manager created. 17:00:18.545 pjsua_core.c .PJSUA state changed: NULL --> CREATED 17:00:18.584 pjsua_core.c .pjsua version 2.11.1 for Linux-6.1.58/aarch64 initialized 17:00:18.633 pjsua_app.c .Turning sound device -99 -99 ON 17:00:18.634 main.c Ready: Success 17:00:18.636 pjsua_app.c .......Call 0 state changed to CALLING

Account list: [ 0] : does not register Online status: Online *[ 1] sip:homeassistant@192.168.1.1:5060: does not register Online status: Online Buddy list: [ 1] <?> sip:+39xxx@192.168.1.1:5060

+=============================================================================+ 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:+39xxx@192.168.1.1:5060 [CALLING]

17:00:18.640 tsx0x7faef956d8 .......Temporary failure in sending Request msg INVITE/cseq=24205 (tdta0x7faef8eaa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT) 17:00:18.669 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) 17:01:08.386 pjsua_app_common.c ... [EARLY] To: sip:+39xxx@192.168.1.1;tag=9EF2DAAB1C6321B2 Call time: 00h:00m:00s, 1st res in 35 ms, conn in 0ms

0 audio G722 @16kHz, sendrecv, peer=192.168.1.1:7078

SRTP status: Not active Crypto-suite: RX pt=9, last update:00h:00m:00.001s ago total 2.0Kpkt 321.9KB (402.4KB +IP hdr) @avg=51.8Kbps/64.7Kbps 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 0.023 0.625 0.000 0.062 TX pt=9, ptime=20, last update:00h:00m:12.829s ago total 1.9Kpkt 309.7KB (387.2KB +IP hdr) @avg=49.8Kbps/62.3Kbps 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 : 1.375 2.484 2.875 2.875 0.610 RTT msec : 0.717 0.805 0.946 0.778 0.086 17:01:08.387 pjsua_app.c .Call 0 is DISCONNECTED [reason=487 (Request Terminated)] 17:01:08.904 sip_transaction.c .....Unable to register CANCEL transaction (key exists) 17:01:08.904 pjsua_call.c ...Failed to send end session message: Object already exists (PJ_EEXISTS) [status=70015] 17:01:09.386 pjsua_app.c ..Turning sound device -99 -99 OFF 17:01:09.945 timer.c .Dumping timer heap: 17:01:09.945 timer.c . Cur size: 1 entries, max: 3070 17:01:09.945 timer.c . Entries: 17:01:09.945 timer.c . _id Id Elapsed Source 17:01:09.945 timer.c . ---------------------------------- 17:01:09.945 timer.c . 4 1 3.959 ../src/pjsua-lib/pjsua_call.c:2848 [Info] Call ended... [Info] Listening for messages via stdin service call...

Ciqsky commented 6 months ago

questa è il mio log dove ho al posto di xxx il mio numero di telefono

Ciqsky commented 6 months ago

sip_parameters: caller_id_uri: sip:homeassistant@192.168.1.1:5060 realm: "*" username: homeassistant password: xxx sip_server_uri: sip:192.168.1.1:5060 pjsua_custom_options: "--no-tcp --ip-addr=192.168.1.105"

questa è la configurazione dove la password è la mia password di accesso e funziona perchè il telefono chiama

Ciqsky commented 6 months ago

service: hassio.addon_stdin data_template: input: call_sip_uri: sip:+39xxx@192.168.1.1:5060 message_tts: Non posso attivare allarme, problema con un sensore data: addon: 89275b70_dss_voip

questo è lo script

sdesalve commented 6 months ago

1 metti screenshot = foto (censura i dati privati non mi interessano) perché se hai messo le opzioni nel campo sbagliato qua perdiamo tempo con le prove e basta

2 FULL addon logs significa TUTTO il log... Dall'avvio alla conclusione della chiamata perché se non mi mandi lo screenshot almeno vedo se vengono rilevate ste benedette opzioni

sdesalve commented 6 months ago

questo è uno screenshot: image

sdesalve commented 6 months ago

[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... �[34m-----------------------------------------------------------�[0m �[34m Add-on: DSS VoIP Notifier�[0m �[34m VoIP Notifier for Home Assistant�[0m �[34m-----------------------------------------------------------�[0m �[34m Add-on version: 4.0.0�[0m �[32m You are running the latest version of this add-on.�[0m �[34m System: Home Assistant OS 10.5 (amd64 / qemux86-64)�[0m �[34m Home Assistant Core: 2023.9.3�[0m �[34m Home Assistant Supervisor: 2023.09.2�[0m �[34m-----------------------------------------------------------�[0m �[34m Please, share the above information when looking for help�[0m �[34m or support in, e.g., GitHub, forums or the Discord chat.�[0m �[34m-----------------------------------------------------------�[0m [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. �[32m[Info] Starting addon...�[0m ### �[33mPJSUA_CUSTOM_OPTIONS = '--no-tcp'�[0m

e questo mi interessa vedere se c'è

Ciqsky commented 6 months ago

image

Ciqsky commented 6 months ago

image

sdesalve commented 6 months ago

Togli questa tanto non serve sip_server_uri: sip:192.168.1.1:5060

controlla che sia disattivata questa opzione image

Ciqsky commented 6 months ago

E' già così

image

sdesalve commented 6 months ago

togli la registrazione tanto non ti serve chiamare Hassio e prova a vedere se il problema si è risolto.

se NO, Fai una chiamata e rispondi dopo 5 secondi Fai una chiamata e rispondi subito

posta ENTRAMBI il log dall'avvio della chiamata in poi

Ciqsky commented 6 months ago

Tolto l'opzione e fa uguale

Test chiamata risposto subito (no audio)

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: Home Assistant OS 11.2 (aarch64 / raspberrypi4-64) Home Assistant Core: 2023.12.4 Home Assistant Supervisor: 2023.12.0

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 --ip-addr=192.168.1.105' [Info] Listening for messages via stdin service call... [Info] Received messages {"call_sip_uri":"sip:+39xxx@192.168.1.1:5060","message_tts":"Non posso attivare allarme, problema con un sensore"} Converting audio file 'http://192.168.1.105:8123/api/tts_proxy/0f1fcd1ded38411701c4934fe3b5527fdb3debb2_it_-_google_translate.mp3'... Audio succesfully converted... Starting SIP Client and calling 'sip:+39xxx@192.168.1.1:5060'... This call will be terminated after '50' seconds. 16:35:16.040 os_core_unix.c !pjlib 2.11.1 for POSIX initialized 16:35:16.042 sip_endpoint.c .Creating endpoint instance... 16:35:16.043 pjlib .select() I/O Queue created (0x7f94609100) 16:35:16.043 sip_endpoint.c .Module "mod-msg-print" registered 16:35:16.043 sip_transport.c .Transport manager created. 16:35:16.043 pjsua_core.c .PJSUA state changed: NULL --> CREATED 16:35:16.084 pjsua_core.c .pjsua version 2.11.1 for Linux-6.1.58/aarch64 initialized 16:35:16.139 pjsua_app.c .Turning sound device -99 -99 ON 16:35:16.139 main.c Ready: Success 16:35:16.141 pjsua_app.c .......Call 0 state changed to CALLING

Account list: [ 0] : does not register Online status: Online *[ 1] sip:homeassistant@192.168.1.1:5060: does not register Online status: Online Buddy list: [ 1] <?> sip:+39xxx@192.168.1.1:5060

+=============================================================================+ 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:+39xxx@192.168.1.1:5060 [CALLING]

16:35:16.146 tsx0x7f938a26d8 .......Temporary failure in sending Request msg INVITE/cseq=15707 (tdta0x7f9389baa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT) 16:35:16.175 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) 16:36:05.870 pjsua_app_common.c ... [EARLY] To: sip:+39xxx@192.168.1.1;tag=C94549091172AD62 Call time: 00h:00m:00s, 1st res in 36 ms, conn in 0ms

0 audio G722 @16kHz, sendrecv, peer=192.168.1.1:7080

SRTP status: Not active Crypto-suite: RX pt=9, last update:00h:00m:00.000s ago total 1.7Kpkt 287.9KB (359.8KB +IP hdr) @avg=46.3Kbps/57.9Kbps 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 0.044 0.500 0.000 0.078 TX pt=9, ptime=20, last update:00h:00m:17.918s ago total 1.9Kpkt 309.6KB (387.0KB +IP hdr) @avg=49.8Kbps/62.3Kbps 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 : 1.125 2.125 2.875 2.875 0.577 RTT msec : 0.595 0.684 0.808 0.701 0.071 16:36:05.871 pjsua_app.c .Call 0 is DISCONNECTED [reason=487 (Request Terminated)] 16:36:06.384 sip_transaction.c .....Unable to register CANCEL transaction (key exists) 16:36:06.384 pjsua_call.c ...Failed to send end session message: Object already exists (PJ_EEXISTS) [status=70015] 16:36:06.871 pjsua_app.c ..Turning sound device -99 -99 OFF 16:36:07.700 timer.c .Dumping timer heap: 16:36:07.700 timer.c . Cur size: 1 entries, max: 3070 16:36:07.700 timer.c . Entries: 16:36:07.700 timer.c . _id Id Elapsed Source 16:36:07.700 timer.c . ---------------------------------- 16:36:07.700 timer.c . 4 1 3.685 ../src/pjsua-lib/pjsua_call.c:2848 [Info] Call ended... [Info] Listening for messages via stdin service call...

Provate altre due volte la seconda dopo 15 secondi e non è andata buon fine (no audio), l'ultima dopo 5 secondi ho risposto e ho sentito l'audio.

You have 1 active call Current call id=0 to sip:+39xxx@192.168.1.1:5060 [CALLING]

16:39:29.157 tsx0x7f96bad6d8 .......Temporary failure in sending Request msg INVITE/cseq=27546 (tdta0x7f96ba6aa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT) 16:39:29.187 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) 16:40:18.506 pjsua_app_common.c ....... [DISCONNCTD] To: sip:+39xxx@192.168.1.1;tag=4BDC2F032E58D8B1 Call time: 00h:00m:00s, 1st res in 35 ms, conn in 0ms

0 audio G722 @16kHz, sendrecv, peer=192.168.1.1:7080

SRTP status: Not active Crypto-suite: RX pt=9, last update:00h:00m:00.000s ago total 2.4Kpkt 394.4KB (493.0KB +IP hdr) @avg=63.9Kbps/79.9Kbps 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 0.047 0.750 0.000 0.084 TX pt=9, ptime=20, last update:00h:00m:02.878s ago total 1.9Kpkt 306.5KB (383.2KB +IP hdr) @avg=49.7Kbps/62.1Kbps 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 : 1.250 2.288 2.875 2.875 0.403 RTT msec : 0.564 0.747 1.571 0.579 0.280 16:40:18.507 pjsua_app.c .....Call 0 is DISCONNECTED [reason=486 (Busy Here)] No current call 16:40:19.506 pjsua_app.c .Turning sound device -99 -99 OFF 16:40:21.009 timer.c .Dumping timer heap: 16:40:21.009 timer.c . Cur size: 0 entries, max: 3070 [Info] Call ended... [Info] Listening for messages via stdin service call... [Info] Received messages {"call_sip_uri":"sip:+39xxx@192.168.1.1:5060","message_tts":"Non posso attivare allarme, problema con un sensore"} Converting audio file 'http://192.168.1.105:8123/api/tts_proxy/0f1fcd1ded38411701c4934fe3b5527fdb3debb2_it_-_google_translate.mp3'... Audio succesfully converted... Starting SIP Client and calling 'sip:+39xxx@192.168.1.1:5060'... This call will be terminated after '50' seconds. 16:40:22.328 os_core_unix.c !pjlib 2.11.1 for POSIX initialized 16:40:22.329 sip_endpoint.c .Creating endpoint instance... 16:40:22.329 pjlib .select() I/O Queue created (0x7f92c6b100) 16:40:22.329 sip_endpoint.c .Module "mod-msg-print" registered 16:40:22.329 sip_transport.c .Transport manager created. 16:40:22.329 pjsua_core.c .PJSUA state changed: NULL --> CREATED 16:40:22.357 pjsua_core.c .pjsua version 2.11.1 for Linux-6.1.58/aarch64 initialized 16:40:22.366 pjsua_app.c .Turning sound device -99 -99 ON 16:40:22.366 main.c Ready: Success 16:40:22.367 pjsua_app.c .......Call 0 state changed to CALLING

Account list: [ 0] : does not register Online status: Online *[ 1] sip:homeassistant@192.168.1.1:5060: does not register Online status: Online Buddy list: [ 1] <?> sip:+39xxx@192.168.1.1:5060

+=============================================================================+ 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:+39xxx@192.168.1.1:5060 [CALLING]

16:40:22.372 tsx0x7f91f046d8 .......Temporary failure in sending Request msg INVITE/cseq=6971 (tdta0x7f91efdaa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT) 16:40:22.402 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress) 16:40:31.324 pjsua_app.c .....Call 0 state changed to CONNECTING 16:40:31.325 pjsua_app_common.c ....... [CONNECTING] To: sip:+39xxx@192.168.1.1;tag=B4AD80A30BBB65F4 Call time: 00h:00m:00s, 1st res in 36 ms, conn in 0ms

0 audio G722 @16kHz, sendrecv, peer=192.168.1.1:7078

SRTP status: Not active Crypto-suite: RX pt=9, last update:00h:00m:00.001s ago total 445pkt 71.2KB (89.0KB +IP hdr) @avg=63.8Kbps/79.7Kbps 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 0.005 0.250 0.000 0.029 TX pt=9, ptime=20, last update:00h:00m:02.437s ago total 363pkt 58.0KB (72.6KB +IP hdr) @avg=52.0Kbps/65.0Kbps 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 : 1.375 1.375 1.375 1.375 0.000 RTT msec : 0.640 0.701 0.762 0.640 0.061

Non è che il problema è la generazione del file mp3?

sdesalve commented 6 months ago

Non è che il problema è la generazione del file mp3?

se clicchi sui link dei file MP3 li riesci a sentire? Anche perché il file è sempre lo stesso

posti la configurazione del TTS?

# Text to speech
tts:
  - platform: google_translate
    service_name: google_translate_say
    language: 'it'
    cache: true
    cache_dir: /config/www/tts
    time_memory: 300
Ciqsky commented 6 months ago

tts:

Ciqsky commented 6 months ago

cache true e cache dir sono commentati da me

sdesalve commented 6 months ago

la cache evita che lo stesso file venga generato più volte. riattivala e controlla che la cartella esista... meglio se la metti sotto www

Ciqsky commented 6 months ago

Allora dopo la modifica 1a chiamata non è andata...la seconda ho aspettato 5/6 secondi è andata le sueccessive due sono andate rispondendo subito alla chiamata: il problema non è il notifier ma la generazione del file MP3. Mi chiedo se c'è un modo per velocizzarla anche perchè se dovessi impostare la generazione del messaggio in modo parametrico ci sarebbero quasi sempre problemi a meno di non aspettare che vengano generate tutte.....problema dell'RPI4?

sdesalve commented 6 months ago

Potrebbe essere la connessione internet lenta o, come dici te, la capacità di calcolo dell'RPi4

A sto punto, genera gli MP3 che ti servono, mettili in www e fagli riprodurre i file mp3 invece di generarli ogni volta

https://github.com/sdesalve/hassio-addons/tree/master/dss_voip#option-audio_file_url-required-if-message_tts-is-not-specified

Ciqsky commented 6 months ago

Grazie per l'aiuto e BUON ANNO!

LucaZiegler commented 2 months ago

Hello @Ciqsky please write in english, we need also a solution. Thx

sdesalve commented 2 months ago

I've suggested to @Ciqsky to use MP3 instead to generate it with TTS .

He hasn't replied if that has resolved this issue with rpi4