Ciao a tutti,
sto provando ad usare questo add on ma non mi riesce di capire dove sbaglio!
Nel log mi viene fuori questo ma, il telefono non scquilla
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...
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Hass.io Add-on: DSS VoIP Notifier
VoIP Notifier for HomeAssistant
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Add-on version: 3.1.5
You are running the latest version of this add-on.
System: Raspbian GNU/Linux 10 (buster) (armv7 / raspberrypi3)
Home Assistant version: 0.105.4
Supervisor version: 200
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Please, share the above information when looking for help
or support in, e.g., GitHub, forums or the Discord chat.
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[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=192.168.178.45'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri": "sip:+393921220XXX1@fritz.box:5060", "message_tts": "Prova messaggio"}
Converting audio file 'https://XXXXXXXXX.duckdns.org:8123/api/tts_proxy/772c970523ed7389b60b2ed1b674b3705d965e37_it_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:+393921220XXX@fritz.box:5060'...
00:10:06.644 os_core_unix.c !pjlib 2.9 for POSIX initialized
00:10:06.645 sip_endpoint.c .Creating endpoint instance...
00:10:06.646 pjlib .select() I/O Queue created (0xae3e78)
00:10:06.646 sip_endpoint.c .Module "mod-msg-print" registered
00:10:06.646 sip_transport. .Transport manager created.
00:10:06.646 pjsua_core.c .PJSUA state changed: NULL --> CREATED
00:10:06.691 pjsua_core.c .pjsua version 2.9 for Linux-4.19.93/armv7l initialized
00:10:06.696 pjsua_app.c .Turning sound device -99 -99 ON
00:10:06.697 main.c Ready: Success
00:10:06.723 pjsua_app.c .......Call 0 state changed to CALLING
Account list:
[ 0] <sip:192.168.178.45:5060>: does not register
Online status: Online
[ 1] <sip:192.168.178.45:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:calas@fritz.box:5060: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:+393921220XXX@fritz.box: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:+393921220XXX@fritz.box:5060 [CALLING]