I’ve tried now and on my NUC + Proxmox + Hassio + FritzBox FreeVoipDeal.com is working:
sip_parameters:
caller_id_uri: sip:[email protected]
realm: '*'
username: aaaaaaaaaaa
password: bbbbbbbbbbb
pjsua_custom_options: '--no-tcp'
My phone is ringing!
Please check your config
base_url Google TTS
Firewall/OS Config
Nothing, i will try to deactivate all other addons. Firewall makes no sense. Google TTS is working fine, i found the file on the system. What do you mean with the point OS Config? I installed the Hass OS image for x86_64.
Ok, here some new informations: I disabled everything, tried so much. Ho into the docker container, pinging freevoipdeal and so on. Nothin helped.
I deinstalled dss_voip and installed dss_voip34. It works directly. Where is the difference between these versions?
Dss_voip34 it’s based on old Hassio add-on’s base images. Like Frank has told me there is no difference and it must work at same way on both versions…
It’s strange. But if it’s working continue to use dss_voip34
Hello sdesalve, thanks for your work!
I have high hopes but I’m stuck with the config and need help. I have a local pbx (mypbx, no external voip provider) and a raspi running home assistant.
The add-on installs, registers, mp3 is created and available on link, but SIP client is not starting and calling the extension ends in voicemail.
this is my config:
sip_parameters:
caller_id_uri: sip:[email protected].(mypbx)
realm: '*'
username: '333'
password: password
sip_server_uri: sip:192.168.(mypbx)
pjsua_custom_options: '--no-tcp --ip-addr=192.168.(raspi)'
I tried various possibilities w/ and w/o port number, w/ and w/o custom options, the result is always the same: no sign of the SIP client starting. Note that I have to put username in quotes since it is numbers only (extension number).
This is the resulting log after running my script:
[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: 3.5.9
You are running the latest version of this add-on
System: Home Assistant OS 6.4 (armv7 / raspberrypi3)
Home Assistant Core: 2021.9.6
Home Assistant Supervisor: 2021.09.0
-----------------------------------------------------------
Please, share the above information when looking for hel
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 service
[services.d] done.
[Info] Starting addon..
PJSUA_CUSTOM_OPTIONS = '--no-tcp --ip-addr=192.168.(raspi)’
[Info] Registering as SIP Client...
PJSUA_CUSTOM_OPTIONS = '--no-tcp --ip-addr=192.168.(raspi)’
-----------------------------------------------------------
SIP Client registered.
Call <sip:[email protected].(mypbx)>/VoIP phone number
to check system status.
You'll find logs in /share/dss_voip/dss_autoanswer.log
-----------------------------------------------------------
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri": "sip:[email protected].(mypbx)”, "message_tts": "Write here your message"}
Converting audio file 'http://homeassistant.local:8123/api/tts_proxy/(blablabla)_en_-_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.
Grazie!
Fail on audio converting. Please check TTS configuration and base_url in your configuration.yaml
Anyone know of a US version of this free VOIP service? Messagenet seems great, but I need a US number.
Ciao, thanks for the pointer, it works now. For those who are running local pbx with self-hosted VoIP (this does not refer to Fritz box):
Hi @VinistoisR
Did you manage to cook this up in a plain docker? I would love to try it but I dont run Hassio.
Thanks.
Basically it’s a bash script.
You can copy it and run passing parameters.
Hassio add-on system and supervisor let me to invoke this script and to pass it chosen parameters.
But I don’t have any plans to rewrite/adapt/fix it for docker. Sorry
SDeSalve
Hello to all.
I am trying to configure this plugin with the provider onsip.com
They provide a free VoIP numbers and Cloud pbx (the VoIP numbers can call one each other for free)…
I can’t set to work correctly…
I have Hassio installed on proxmox into a Nuc
My fiber provider is Telecom Italia with another working phone number (also via VoIP). Maybe Is this the problem? I noticed that it’s VoIP settings also use the 5060 port…
Please post configuration and full addon logs
Redact your private info
Hi @sdesalve
thanks for your quick interest!
Here it is my configuration (same char => same word)
caller_id_uri: sip:[email protected]
realm: yyyyyyyyyyyy.onsip.com
username: yyyyyyyyyyyy_xxxxxx_1
password: jfnwnfsnfsjdnfksjnf
pjsua_custom_options: '--outbound=sip:yyyyyyyyyyyy.onsip.com --no-tcp'
Here is the log:
[Info] Received messages {"call_sip_uri": "sip:[email protected]", "message_tts": "Test"}
Converting audio file 'https://bbbbbbbb.duckdns.org/api/tts_proxy/yuikyuiyuiyu_it_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '50' seconds.
09:21:07.366 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
09:21:07.370 sip_endpoint.c .Creating endpoint instance...
09:21:07.373 pjlib .select() I/O Queue created (0x7ff3b36bc100)
09:21:07.373 sip_endpoint.c .Module "mod-msg-print" registered
09:21:07.373 sip_transport.c .Transport manager created.
09:21:07.373 pjsua_core.c .PJSUA state changed: NULL --> CREATED
09:21:07.455 pjsua_core.c .pjsua version 2.11.1 for Linux-5.10.75/x86_64 initialized
09:21:07.515 pjsua_app.c .Turning sound device -99 -99 ON
09:21:07.515 main.c Ready: Success
09:21:08.289 tsx0x7ff3b352ac18 ....Temporary failure in sending Request msg INVITE/cseq=16193 (tdta0x7ff3b3524aa8), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
09:21:08.289 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:[email protected]: 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]
>>> 09:21:08.515 pjsua_app.c .Turning sound device -99 -99 OFF
09:21:40.290 pjsua_app.c ....Call 0 is DISCONNECTED [reason=408 (Request Timeout)]
No current call
>>> 09:21:59.006 timer.c .Dumping timer heap:
09:21:59.006 timer.c . Cur size: 0 entries, max: 3070
[Info] Call ended...
[Info] Listening for messages via stdin service call...
Here the configuration info from onsip.com (same color => same word)
Thank you!
First part of addon logs? So we can view if option are recognized.
Try also to specify 5060 port after server and right port after proxy
Here is the first part:
[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: 3.5.9
You are running the latest version of this add-on.
System: Home Assistant OS 6.6 (amd64 / qemux86-64)
Home Assistant Core: 2021.11.5
Home Assistant Supervisor: 2021.10.8
-----------------------------------------------------------
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 = '--outbound=sip:yyyyyyyyyyyy.onsip.com --no-tcp'
[Info] Listening for messages via stdin service call...
I’ve also already try with :5060 but with no success…
for future reference
sip_parameters:
caller_id_uri: sip:[email protected]:5060
realm: '*'
username: Auth Username
password: SIP Password
pjsua_custom_options: '--outbound=sip:sip.onsip.com:5060;lr --no-tcp'
I hadn’t realised this before but if I understand correctly the audio starts playing the moment the call is initiated right? But there’s no way for the audio to start only after you answer?
No unfortunately.
If you make an audio shorted than call duration it will be repeated until call end
Hi,
I’m trying to use that addon with a local pbx (Panasonic) with SIP extensions. I could connect a Doorbird doorbell without problems.
Here, I tried with many different pjsua_custom_options but without success.
The logs says that it is connected but it’s not the case.
Any idea ?