There isn’t any “list of valid VoIP services”. You can use any VoIP provider. There is some examples for already used providers… Feel free to try whatever service you like… Use examples for configuration that will work with your choice
You can use any TTS service that is recognised by Hassio… See docs
I have tested both Google (did not specify ant TTS so the default is used) and Microsoft.
There is no phone call going out at all and I see no error in the HA log.
I use pushover so notification do work. I did not change anything since I have posted my working solution.
I have no idea what seems to be the problem.
This is inaccurate, from the last update Google Translate also supports Hebrew, the language setting is ‘iw’
If you’ll need help, you must post
Addon logs, your host type and more details.
Otherwise I can’t help nobody. I can’t see anything at this distance… It’s far far away and my eyeglasses aren’t enough 
Have you tried dss_voip34 version?
Got it.
No.
What is new there?
It is working now although I can’t call myself. If I want the system to send alarms, I must use my home phone and call my mobile.
Sorry I don’t understand the point.
My VoIP service allow me to call all phone numbers, even mobile and landline
Try other VoIP provider
Obviously you can’t call from a VoIP extension same extension… You’ll need another one
Going back to dss_voip34. What is new in this one?
@sdesalve
I really enjoy your plugin, and use it a lot,
I thank you very much,
Today the plugin dials me and plays me the message I submitted,
Is there an option to hear the message, when I dial the number specified in the extension,
Thank you
But only a dummy MP3 file will be played…
I’m looking for that option, because sometimes I get the alerts while I’m in a meeting and I can not answer the call,
So I thought that if it was possible to dial into a number and hear the alert it would be fantastic,
But I understand from you that there is no such possibility, maybe in the next versions it will be …
Thank you for the wonderful plugin
@sdesalve any tips on how to set up this with fastweb? and can you explain how to specify the numbwer to call an how to set up a script with an example?
What is fastweb?
Google helpfully tells me it will find me College Scholarships.
Do you have all VoIP parameters?
Can you successfully place a call thru your computer with microsip.org client phone?
To use this (and all other add-ons) you must invoke it in your automations/scripts… As is stated here
yes i have all the parameter and i’ve placed a test call an it worked but i wasn’t able to place a call with home assistsnt for some reason
it’s an italian isp
Please be accurate, otherwise I can’t help you.
Non ho la sfera di cristallo 
e non posso rispondere alla domanda “any tips on how to set up this with fastweb” senza quanto richiesto
bye
you’re right but i didn’t know what you might need that’s why i hopted for a generic: “any tips on how to set up this with fastweb”
1 yes i placed the test call with microsip.oeg
2
lan specs
single vlan with router ip 192.168.50.1 subnet mask 255.255.255.0
add-on config
caller_id_uri: sip:[email protected]
realm: ‘*’
username: ‘xxxx’
password: xxx
[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…
after call
[Info] Received messages {“call_sip_uri”:“sip:mydadnumber (tried with and without national prefix)@sip.fastwebnet.it”,“message_tts”:“ciao”}
Converting audio file ‘https://myHa external address/api/tts_proxy/1e4e888ac66f8dd41e00c5a7ac36a32a9950d271_it_-_google_translate.mp3’…
Audio succesfully converted…
Starting SIP Client and calling ‘sip:mydadnumber (tried with and without national prefix)@sip.fastwebnet.it’…
This call will be terminated after ‘50’ seconds.
20:22:01.516 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
02:50:17.517 sip_endpoint.c .Creating endpoint instance…
02:50:17.517 pjlib .select() I/O Queue created (0xb66b40c8)
02:50:17.517 sip_endpoint.c .Module “mod-msg-print” registered
02:50:17.517 sip_transport. .Transport manager created.
02:50:17.517 pjsua_core.c .PJSUA state changed: NULL → CREATED
22:19:05.535 pjsua_core.c .pjsua version 2.11.1 for Linux-5.10.63/armv7l initialized
14:08:41.541 pjsua_app.c .Turning sound device -99 -99 ON
14:08:41.541 main.c Ready: Success
17:58:51.564 tsx0xb652d074 …Failed to send Request msg INVITE/cseq=13018 (tdta0xb6530454)! err=70018 (gethostbyname() has returned error (PJ_ERESOLVE))
17:58:51.564 pjsua_app.c …Call 0 is DISCONNECTED [reason=502 (gethostbyname() has returned error (PJ_ERESOLVE))]
17:58:51.564 pjsua_app.c .Turning sound device -99 -99 OFF
Account list:
[ 0] sip:172.30.33.6:5060: does not register
Online status: Online
[ 1] sip:172.30.33.6:5060;transport=TCP: does not register
Online status: Online
*[ 2] sip:xxx(my user)@sip.fastwebnet.it: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:mydadnumber (tried with and without national prefix)@sip.fastwebnet.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 0 active call
my script
service: hassio.addon_stdin
data_template:
addon: 89275b70_dss_voip
input:
call_sip_uri: sip:mydadnumber (tried with and without national prefix)@sip.fastwebnet.it
message_tts: ciao
i think i didn’t forgot anything if you need anything else please let me know
thanks for the help
And
Did you haven’t installed version for ARM processors?