Ok, I think did all as per instructions but receive this error
PJSUA_CUSTOM_OPTIONS = '--outbound=sip:sip.messagenet.it:5061;lr'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri": "sip:[email protected]", "message_tts": "Attenzione e' scattato l'allarme in via xxxxx"}
Converting audio file 'https://xxx.duckdns.org:8123/api/tts_proxy/xxxx5d3b51a423b0b2e7282xxxxxx_it_-_google_translate.mp3'...
[cmd] /run.sh exited 2
[cont-finish.d] executing container finish scripts...
[cont-finish.d] 99-message.sh: executing...
-----------------------------------------------------------
Oops! Something went wrong.
We are so sorry, but something went terribly wrong when
starting or running this add-on.
Be sure to check the log above, line by line, for hints.
-----------------------------------------------------------
[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.
What kind of system are you using?
I’ve tested it on ResinOS/HassOS on Raspberry 3b.
Other users have installed it on Debian+Docker+Hassio and Ubuntu+Docker+Hassio on Intel NUC without problems…
Should be compatible with everything, but let me know other details about your setup…
Cannot seem to get this to work with Linphone so far. I think it just comes down to the fact that i’m not especially experienced when it comes to sip/voip configuration. So if anyone gets it to work with Linphone in the future, i would gladly appreciate any posts with configs!
Hass.io Add-on: DSS VoIP Notifier
VoIP Notifier for HomeAssistant
-----------------------------------------------------------
Add-on version: 2.0.1
You are running the latest version of this add-on.
System: HassOS 1.13 (amd64 / qemux86-64)
Home Assistant version: 0.98.1
Supervisor version: 184
-----------------------------------------------------------
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] Registering as SIP Client...
-----------------------------------------------------------
SIP Client registered.
Call <sip:[email protected]>/VoIP phone number
to check system status.
You'll find logs in /share/dss_voip/dss_pjsua_autoanswer.log
-----------------------------------------------------------
[Info] Listening for messages via stdin service call...
So it seems to register but when i make this service call
- service: hassio.addon_stdin
data_template:
addon: 89275b70_dss_voip
input: {"call_sip_uri":"sip:[email protected]","message_tts":"Welcome, the door is unlocked"}
This happens:
16:40:42.244 os_core_unix.c !pjlib 2.8 for POSIX initialized
16:40:42.244 sip_endpoint.c .Creating endpoint instance...
16:40:42.245 pjlib .select() I/O Queue created (0x5646d0209a70)
16:40:42.245 sip_endpoint.c .Module "mod-msg-print" registered
16:40:42.245 sip_transport.c .Transport manager created.
16:40:42.245 pjsua_core.c .PJSUA state changed: NULL --> CREATED
16:40:42.309 pjsua_core.c .pjsua version 2.8 for Linux-4.14.82/x86_64 initialized
16:40:42.310 pjsua_app.c .Turning sound device -99 -99 ON
16:40:42.310 main.c Ready: Success
16:40:42.324 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
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:[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]
>>> 16:40:42.356 pjsua_app.c .....Call 0 is DISCONNECTED [reason=407 (Proxy Authentication Required)]
16:40:42.356 pjsua_app_common.c .....
[DISCONNCTD] To: sip:[email protected]
Call time: 00h:00m:00s, 1st res in 46 ms, conn in 0ms
16:40:43.311 pjsua_app.c .Turning sound device -99 -99 OFF
[Info] Call ended...
[Info] Listening for messages via stdin service call...
It does not connect to my intercom, seems to be a proxy error right?
SIP Client should register correctly. I need to see dss_pjsua_autoanswer.log file in your /share/dss_voip folder.
Try to call [email protected] to test if it has been registered
As you stated error is related with proxy auth
pjsua_app.c …Call 0 is DISCONNECTED [reason=407 (Proxy Authentication Required)]
I’m searching PjSua option to activate it, if possible
