i’ve tried to install that version but it gives me this error
‘Connection aborted.’, ConnectionResetError(104, 'Connection reset by peer
Full logs…
how can igive you any logs if when i click istall it won’t let me? that’s the message that appears when i click on the istall button
I’ve not understood that error was during setup… Was not specified, sorry
Post your hassio Supervisor error logs and info about your hassio setup… Do you have fully supported hassio? Raspbian or anything else?
With ARM processors you must use arm version… Otherwise it will not place any call
Try to update to the last hassio/hassos version, reboot your host machine and try to install
sorry if i came of rude it was not my intention
anyway this is the line that appear in the supervisor log when i try to istall the arm version
22-07-27 00:13:51 ERROR (SyncWorker_4) [supervisor.utils.json] Can’t read json from /data/addons/git/89275b70/dss_voipARM/build.json: Expecting property name enclosed in double quotes: line 7 column 5 (char 213)
22-07-27 00:13:51 INFO (SyncWorker_4) [supervisor.docker.addon] Starting build for 89275b70/armv7-addon-dss_voipARM:4.0.0
22-07-27 00:13:52 ERROR (SyncWorker_4) [supervisor.docker.addon] Can’t build 89275b70/armv7-addon-dss_voipARM:4.0.0: (‘Connection aborted.’, ConnectionResetError(104, ‘Connection reset by peer’))
22-07-27 00:14:04 ERROR (SyncWorker_5) [supervisor.utils.json] Can’t read json from /data/addons/git/89275b70/dss_voipARM/build.json: Expecting property name enclosed in double quotes: line 7 column 5 (char 213)
22-07-27 00:14:04 INFO (SyncWorker_5) [supervisor.docker.addon] Starting build for 89275b70/armv7-addon-dss_voipARM:4.0.0
22-07-27 00:14:05 ERROR (SyncWorker_5) [supervisor.docker.addon] Can’t build 89275b70/armv7-addon-dss_voipARM:4.0.0: (‘Connection aborted.’, ConnectionResetError(104, ‘Connection reset by peer’))
i’ve an home assistant supervised install on top of raspbian buster (2022.7.7)
i’ve rebooted the host like you suggested but nothing has changed
Quite possibly your problem is that buster is not supported, only bullseye. Nor is raspbian officially supported, only debian. Having said that I have seen many reports that raspbian does work, but of course you’d need to go up to bullseye.
Thank you for the add-on. However, for some reason, it’s not working for me. The log shows that everything is working but I’m not getting the 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: 4.0.0
You are running the latest version of this add-on.
System: Home Assistant OS 8.4 (amd64 / qemux86-64)
Home Assistant Core: 2022.8.1
Home Assistant Supervisor: 2022.07.0
-----------------------------------------------------------
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...
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:110@myipis here:5060","message_tts":"Write here your message"}
Converting audio file 'http://HAip:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:110@myipis here:5060'...
This call will be terminated after '50' seconds.
18:55:50.871 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
18:55:50.875 sip_endpoint.c .Creating endpoint instance...
18:55:50.876 pjlib .select() I/O Queue created (0x7f651a0e7100)
18:55:50.876 sip_endpoint.c .Module "mod-msg-print" registered
18:55:50.876 sip_transport.c .Transport manager created.
18:55:50.876 pjsua_core.c .PJSUA state changed: NULL --> CREATED
18:55:50.917 pjsua_core.c .pjsua version 2.11.1 for Linux-5.15.55/x86_64 initialized
18:55:50.934 pjsua_app.c .Turning sound device -99 -99 ON
18:55:50.934 main.c Ready: Success
18:55:50.945 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.3:5060>: does not register
Online status: Online
[ 1] <sip:172.30.33.3:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:110@myipis here:5060: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:110@myipis here: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:110@myipis here :5060 [CALLING]
System Information
| version | core-2022.8.1 |
|---|---|
| installation_type | Home Assistant OS |
| dev | false |
| hassio | true |
| docker | true |
| user | root |
| virtualenv | false |
| python_version | 3.10.5 |
| os_name | Linux |
| os_version | 5.15.55 |
| arch | x86_64 |
| timezone | Asia/Riyadh |
| config_dir | /config |
Home Assistant Community Store
| GitHub API | ok |
|---|---|
| GitHub Content | ok |
| GitHub Web | ok |
| GitHub API Calls Remaining | 4599 |
| Installed Version | 1.26.2 |
| Stage | running |
| Available Repositories | 1070 |
| Downloaded Repositories | 8 |
Home Assistant Cloud
| logged_in | true |
|---|---|
| subscription_expiration | January 1, 2018, 3:00 AM |
| relayer_connected | false |
| remote_enabled | false |
| remote_connected | false |
| alexa_enabled | true |
| google_enabled | true |
| remote_server | null |
| can_reach_cert_server | ok |
| can_reach_cloud_auth | ok |
| can_reach_cloud | ok |
Home Assistant Supervisor
| host_os | Home Assistant OS 8.4 |
|---|---|
| update_channel | stable |
| supervisor_version | supervisor-2022.07.0 |
| agent_version | 1.2.1 |
| docker_version | 20.10.14 |
| disk_total | 30.8 GB |
| disk_used | 6.3 GB |
| healthy | true |
| supported | true |
| board | ova |
| supervisor_api | ok |
| version_api | ok |
| installed_addons | File editor (5.3.3), Cloudflared (2.0.6), Samba share (10.0.0), Mosquitto broker (6.1.2), DSS VoIP Notifier (4.0.0) |
Dashboards
| dashboards | 2 |
|---|---|
| resources | 2 |
| views | 3 |
| mode | storage |
Recorder
| oldest_recorder_run | July 27, 2022, 10:45 AM |
|---|---|
| current_recorder_run | August 5, 2022, 10:30 AM |
| estimated_db_size | 793.36 MiB |
| database_engine | sqlite |
| database_version | 3.38.5 |
what’s “myipis here”? here you must set your account SIP server address/IP
please add base_url with your public HA URL
If SIP address of VoIP is correct and you don’t have setted in that parameter you hassio IP, try also
pjsua_custom_options: '--no-tcp'
I redacted The IP addresses here is the log after adding pjsua_custom_options: '--no-tcp'
[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: Home Assistant OS 8.4 (amd64 / qemux86-64)
Home Assistant Core: 2022.8.1
Home Assistant Supervisor: 2022.07.0
-----------------------------------------------------------
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'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:[email protected]:5060","message_tts":"Write here your message"}
Converting audio file 'http://10.18.100.231:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]:5060'...
This call will be terminated after '50' seconds.
21:24:52.909 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
21:24:52.910 sip_endpoint.c .Creating endpoint instance...
21:24:52.910 pjlib .select() I/O Queue created (0x7fc2a182e100)
21:24:52.910 sip_endpoint.c .Module "mod-msg-print" registered
21:24:52.910 sip_transport.c .Transport manager created.
21:24:52.910 pjsua_core.c .PJSUA state changed: NULL --> CREATED
21:24:52.924 pjsua_core.c .pjsua version 2.11.1 for Linux-5.15.55/x86_64 initialized
21:24:52.928 pjsua_app.c .Turning sound device -99 -99 ON
21:24:52.928 main.c Ready: Success
21:24:52.929 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.3:5060>: does not register
Online status: Online
*[ 1] sip:[email protected]:5060: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:[email protected]: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:[email protected]:5060 [CALLING]
>>> 21:24:53.928 pjsua_app.c .Turning sound device -99 -99 OFF
21:25:24.929 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 21:25:44.442 timer.c .Dumping timer heap:
21:25:44.442 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...
and this is it without it
[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: Home Assistant OS 8.4 (amd64 / qemux86-64)
Home Assistant Core: 2022.8.1
Home Assistant Supervisor: 2022.07.0
-----------------------------------------------------------
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...
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:[email protected]:5060","message_tts":"Write here your message"}
Converting audio file 'http://10.18.100.231:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]:5060'...
This call will be terminated after '50' seconds.
21:26:38.350 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
21:26:38.350 sip_endpoint.c .Creating endpoint instance...
21:26:38.351 pjlib .select() I/O Queue created (0x7fb544eba100)
21:26:38.351 sip_endpoint.c .Module "mod-msg-print" registered
21:26:38.351 sip_transport.c .Transport manager created.
21:26:38.351 pjsua_core.c .PJSUA state changed: NULL --> CREATED
21:26:38.364 pjsua_core.c .pjsua version 2.11.1 for Linux-5.15.55/x86_64 initialized
21:26:38.369 pjsua_app.c .Turning sound device -99 -99 ON
21:26:38.369 main.c Ready: Success
21:26:38.369 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.3:5060>: does not register
Online status: Online
[ 1] <sip:172.30.33.3:5060;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]: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:[email protected]:5060 [CALLING]
>>> 21:26:39.370 pjsua_app.c .Turning sound device -99 -99 OFF
21:27:10.370 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 21:27:29.879 timer.c .Dumping timer heap:
21:27:29.879 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...
And both times this was my serves call
service: hassio.addon_stdin
data: {
addon: 89275b70_dss_voip,
input: {"call_sip_uri":"sip:[email protected]:5060","message_tts":"Write here your message"}
}
By the way when I monitor my 3cx admin panel I noticed there is no calls being made.
Your local Lan is 10.18.100.*?
Your HA IP is 10.18.100.231?
Your local 3cx server is 10.18.100.157?
yes to all
and I just remembered I’m using port 4060 because it conflict with my STC (ISP provider) router.
in my Grandstream HT813 I’m using UDP
the log after editing the port
[cont-init.d] done.
[services.d] starting services
[services.d] done.
[Info] Starting addon...
PJSUA_CUSTOM_OPTIONS = '--no-tcp'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:[email protected]:4060","message_tts":"Write here your message"}
Converting audio file 'http://10.18.100.231:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]:4060'...
This call will be terminated after '50' seconds.
22:16:18.748 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
22:16:18.749 sip_endpoint.c .Creating endpoint instance...
22:16:18.750 pjlib .select() I/O Queue created (0x7f292cb84100)
22:16:18.750 sip_endpoint.c .Module "mod-msg-print" registered
22:16:18.750 sip_transport.c .Transport manager created.
22:16:18.750 pjsua_core.c .PJSUA state changed: NULL --> CREATED
22:16:18.766 pjsua_core.c .pjsua version 2.11.1 for Linux-5.15.55/x86_64 initialized
22:16:18.773 pjsua_app.c .Turning sound device -99 -99 ON
22:16:18.773 main.c Ready: Success
22:16:18.775 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.3:5060>: does not register
Online status: Online
*[ 1] sip:[email protected]:4060: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:[email protected]:4060
+=============================================================================+
| 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]:4060 [CALLING]
>>> 22:16:18.849 tsx0x7f292c9f36d8 .......Temporary failure in sending Request msg INVITE/cseq=2547 (tdta0x7f292c9ecaa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
22:16:19.118 pjsua_app.c .....Call 0 state changed to EARLY (180 Ringing)
22:16:19.118 pjsua_app.c .....Call 0 state changed to CONNECTING
22:16:19.123 pjsua_app.c .....Call 0 state changed to CONFIRMED
22:16:19.416 pjsua_app_common.c .......
[CONFIRMED] To: sip:[email protected];tag=c61a1115
Call time: 00h:00m:00s, 1st res in 345 ms, conn in 350ms
#0 audio speex @8kHz, sendrecv, peer=-
SRTP status: Not active Crypto-suite:
RX pt=97, last update:00h:00m:00.000s 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=97, ptime=20, last update:never
total 15pkt 570B (1.1KB +IP hdr) @avg=15.3Kbps/31.4Kbps
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
22:16:22.994 pjsua_app_common.c ........
[DISCONNCTD] To: sip:[email protected];tag=c61a1115
Call time: 00h:00m:03s, 1st res in 345 ms, conn in 350ms
#0 audio speex @8kHz, sendrecv, peer=-
SRTP status: Not active Crypto-suite:
RX pt=97, last update:00h:00m:00.000s 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=97, ptime=20, last update:never
total 86pkt 3.0KB (6.4KB +IP hdr) @avg=6.8Kbps/14.4Kbps
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
22:16:22.994 pjsua_app.c ......Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
22:16:23.994 pjsua_app.c .Turning sound device -99 -99 OFF
No current call
>>> 22:17:10.285 timer.c .Dumping timer heap:
22:17:10.285 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...
Has ringed? Did you hear anything when pickup receiver?
didn’t even ring and nothing in 3cx log or call record
update
it’s working. it was the port. Then I realized that I was calling the same extension 
Thank you
So I use my home assistant google tts in English and Arabic. If you want to use other languages, you can jus keep adding Google TTS intros just like this
tts:
- platform: google_translate
- platform: google_translate
service_name: google_translate_say_ar
language: 'ar'
- platform: google_translate
service_name: google_translate_say_sp
language: 'sp'
and write your message in that languages keyboard layout
Arabic message
service: hassio.addon_stdin
data: {
addon: 89275b70_dss_voip,
input: {"call_sip_uri":"sip:[email protected]:4060","message_tts":"مرحباً بكم"}
}
Why can’t I install the ARM version? It always gives:
The normal (non-ARM) version instead installs just fine.
I’m running on a Raspberry Pi 4 - so I thought I better take the ARM version…
Try the normal version.
Sometimes on raspberry call will not be placed with new addon base image. So I left old addon base image and make an ARM Version…
so i’m not the only one with this problem
@sdesalve sorry to bother you again but i didn’t understood one thing the problem with the non-arm version istalled on arm causes issues sometimes or everytime? because from the response you gave me i assumed it was everytime
Yes, on some system the called extension will not ring and call will not be placed
I have been able to get everything working: my configuration is HA in container, with fritzbox, and the tests with audio_file_url are successful.
Now I’d like to change it using tts picotts: I tried that, but when it should make the call, the integration crashes (and the watchdog restarts it).
At the moment I’m using google_say, but I don’t know if this is my problem or if dss_voip can’t work with picotts.
Thanks anyway for this great addon!!
Configs:
configuration.yaml
# Text to speech
tts:
- platform: picotts
language: "it-IT"
DSS_Voip_notifier:
sip_parameters:
caller_id_uri: sip:[email protected]
realm: "*"
username: voip-test1
password: test_voip1
pjsua_custom_options: "--no-tcp --ip-addr=192.168.1.100"
max_call_time: 20
Automation:
alias: Test telefonata
description: ""
trigger:
- platform: state
entity_id:
- input_boolean.prevent_deep_sleep
condition: []
action:
- service: hassio.addon_stdin
data_template:
addon: 89275b70_dss_voip
input:
call_sip_uri: sip:392*******@192.168.1.1:5060
message_tts: Prova segnalazione allarme