Hi,
I have a little problem and hope you can help me 
The call is running but I get no voice.
In the logs I can see the follow states :
[Info] Received messages {“call_sip_uri”: “sip:*[email protected]”, “message_tts”: “Test alarm”}
Converting audio file ‘http://172.168.45.100:8123/api/tts_proxy/66884dc6_de_-_google_translate.mp3’…
Audio succesfully converted…
Any idea why the voice not working at the call?
Did you have specified base_url in google tts configuration.yaml?
Yes, this url is there specified.
Have you specified
tts:
- platform: google_translate
service_name: google_translate_say
language: 'it'
cache: true
cache_dir: /config/www/tts
time_memory: 300
base_url: http://172.168.45.10:8123
Is your LAN ip range 172.168.45.1/254? Or 172.168.45.10 is docker IP of your hassio?
Can you hear anything clicking on ‘http://172.168.45.100:8123/api/tts_proxy/66884dc6_de_-_google_translate.mp3’…?
tts:
- platform: google_translate
service_name: google_translate_say
language: ‘it’
cache: true
cache_dir: /config/www/tts
time_memory: 300
base_url: http://172.168.45.10:8123
yes
Is your LAN ip range 172.168.45.1/254?
Yes. Home Assistant is running on Virtual Machine with the IP 172.168.45.100
Can you hear anything clicking on ‘http://172.168.45.100:8123/api/tts_proxy/66884dc6_de_-_google_translate.mp3’…?
Yes, when I open this link in the browser I can hear the voice with the text “Test alarm”
try
pjsua_custom_options: ‘–no-tcp’
if not working please post full log…
Hi, i have a problem with fritzbox configuration.
This is my config:
sip_parameters:
caller_id_uri: 'sip:[email protected]:5060'
realm: '*'
username: '+39telephonenumber'
password: 'password of my voip line'
pjsua_custom_options: '--ip-addr=ip_del_raspberry'
execute command, have this log and add-on stop working
[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: 3.5.7
You are running the latest version of this add-on.
System: Home Assistant OS 5.13 (armv7 / raspberrypi3)
Home Assistant Core: 2021.5.0
Home Assistant Supervisor: 2021.04.3
-----------------------------------------------------------
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 = '--ip-addr=ip_del_raspberry'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri": "sip:[email protected]", "message_tts": "Prova messaggio"}
Converting audio file 'https://xxxxxxxx.net:8123/api/tts_proxy/772c970523ed7389b60b2ed1b674b3705d965e37_it_-_google_translate.mp3'...
[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.
if you click this, can you ear anything?
Check hassio-addons/dss_voip at master · sdesalve/hassio-addons · GitHub
and add
base_url: Please provide a valid base url for Google TTS
to your Google TTS Config
Correct the base url error, now have this in log, and no call.
[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: 3.5.7
You are running the latest version of this add-on.
System: Home Assistant OS 5.13 (armv7 / raspberrypi3)
Home Assistant Core: 2021.5.0
Home Assistant Supervisor: 2021.04.3
-----------------------------------------------------------
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 = '--ip-addr=192.x.x.x'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri": "sip:[email protected]", "message_tts": "Prova messaggio"}
Converting audio file 'http://xxxxi.xxx.net:8123/api/tts_proxy/772c970523ed7389b60b2ed1b674b3705d965e37_it_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '50' seconds.
22:53:09.669 os_core_unix.c !pjlib 2.9 for POSIX initialized
05:21:25.670 sip_endpoint.c .Creating endpoint instance...
10:49:41.671 pjlib .select() I/O Queue created (0x766b20b8)
10:49:41.671 sip_endpoint.c .Module "mod-msg-print" registered
10:49:41.671 sip_transport. .Transport manager created.
10:49:41.671 pjsua_core.c .PJSUA state changed: NULL --> CREATED
12:08:37.712 pjsua_core.c .pjsua version 2.9 for Linux-5.4.83/armv7l initialized
15:01:41.716 pjsua_app.c .Turning sound device -99 -99 ON
15:01:41.716 main.c Ready: Success
18:02:45.750 tsx0x7652a094 ....Failed to send Request msg INVITE/cseq=2517 (tdta0x7652d474)! err=70018 (gethostbyname() has returned error (PJ_ERESOLVE))
18:02:45.750 pjsua_app.c .......Call 0 is DISCONNECTED [reason=502 (gethostbyname() has returned error (PJ_ERESOLVE))]
18:02:45.750 pjsua_app_comm .......
[DISCONNCTD] To: sip:[email protected]
Call time: 00h:00m:00s, 1st res in 33 ms, conn in 0ms
>>>>
Account list:
[ 0] <sip:192.x.x.x:5060>: does not register
Online status: Online
[ 1] <sip:192.x.x.x:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:[email protected]:5060: 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 0 active call
please post your
LAN IP
Raspberry IP
Addon Config
WAIT!
no, you must create an account for a SIP Client on your Fritz.
Please read this and follow all steps
If you don’t speak Italian, the GitHub guide is just as clear
Questa parte non è per niente chiara. Riesci a spiegarmi meglio cosa devo fare sul Fritz?
Ok, I think I have a similar issue like the previous poster. My add-on was working nicely until I upgraded HA to core-2021.5.3. Calls are still placed. but I don’t hear anything. Neither if I try tts nor audio files.
[Info] Received messages {"call_sip_uri": "sip:**[email protected]", "message_tts": "Bla bla bla. Danke.", "call_duration": "20"}
CALL_DURATION = '20'
Converting audio file 'https://xxx.duckdns.org/api/tts_proxy/2c455049b61debda7d9b2af346a56800408a3169_de_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:**[email protected]'...
This call will be terminated after '20' seconds.
16:16:40.608 os_core_unix.c !pjlib 2.9 for POSIX initialized
16:16:40.609 sip_endpoint.c .Creating endpoint instance...
16:16:40.609 pjlib .select() I/O Queue created (0x7f9d9370f0)
16:16:40.609 sip_endpoint.c .Module "mod-msg-print" registered
16:16:40.609 sip_transport.c .Transport manager created.
16:16:40.609 pjsua_core.c .PJSUA state changed: NULL --> CREATED
16:16:40.629 pjsua_core.c .pjsua version 2.9 for Linux-5.4.83/aarch64 initialized
16:16:40.632 pjsua_app.c .Turning sound device -99 -99 ON
16:16:40.632 main.c Ready: Success
16:16:40.634 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.9:5060>: does not register
Online status: Online
[ 1] <sip:172.30.33.9:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:[email protected]:5060: 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]
>>> 16:16:40.644 pjsua_app.c SIP TCP transport is connected to 192.168.178.1:5060
16:16:40.711 pjsua_app.c .....Call 0 state changed to EARLY (183 Session Progress)
16:16:42.680 pjsua_app.c .....Call 0 state changed to CONNECTING
16:16:42.681 pjsua_app.c .....Call 0 state changed to CONFIRMED
>>> 16:17:00.602 pjsua_app.c .....Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
16:17:00.602 pjsua_app_common.c .....
[DISCONNCTD] To: sip:**[email protected];tag=FF27E9562D6847EC
Call time: 00h:00m:17s, 1st res in 79 ms, conn in 2049ms
#0 audio G722 @16kHz, sendrecv, peer=-
SRTP status: Not active Crypto-suite:
RX pt=9, last update:00h:00m:09.585s ago
total 0pkt 0B (0B +IP hdr) @avg=0bps/0bps
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 0.000 0.000 0.000 0.000
TX pt=9, ptime=20, last update:never
total 784pkt 125.4KB (156.8KB +IP hdr) @avg=50.4Kbps/63.0Kbps
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 : 0.000 0.000 0.000 0.000 0.000
RTT msec : 0.000 0.000 0.000 0.000 0.000
16:17:01.603 pjsua_app.c ..Turning sound device -99 -99 OFF
[Info] Call ended...
[Info] Listening for messages via stdin service call...
try sip:**[email protected]:5060
seems that base_url was properly configured so try to disable tcp transport
pjsua_custom_options: '--no-tcp'
Thanks for the quick reply - didn’t work. I changed from
sip_parameters:
caller_id_uri: 'sip:[email protected]:5060'
realm: '*'
username: homeassistant
password: '<replace-me-with-digits-only>'
pjsua_custom_options: '--ip-addr=192.168.178.83'
to
sip_parameters:
caller_id_uri: 'sip:[email protected]:5060'
realm: '*'
username: homeassistant
password: '<replace-me-with-digits-only>'
pjsua_custom_options: '--no-tcp'
and I also
addon: 89275b70_dss_voip
input:
call_sip_uri: 'sip:**[email protected]:5060'
audio_file_url: 'https://***.duckdns.org/local/open_door_short.mp3'
call_duration: '10'
with no luck. Call is placed, but when I pickup the phone … silence.
Can I change the volume somehow?
Best, Stefan
sdesalve, you da man! two leading spaces make the world 
This works now …
sip_parameters:
caller_id_uri: 'sip:[email protected]:5060'
realm: '*'
username: homeassistant
password: '***'
pjsua_custom_options: '--ip-addr=192.168.178.83'
the script uses the :5060
addon: 89275b70_dss_voip
input:
call_sip_uri: 'sip:**[email protected]:5060'
message_tts: Bla Bla Bla. Danke.
call_duration: '15'
and everything works
finally it works ok, its amazing for security and automations, but i have an issue. I have to call the service twice.
First call and 4-5 later a second service call, and then make the call. I have set my automations with that, two calls to the service.
Is add-on running first time you call add-on? Have you checked start at boot option?