Guten Abend zusammen.
Ich bin ganz neu hier, habe aber hier wohl gleichgesinnte gefunden.
Konkret geht's mir um das dss_voip add-on.
Ich habe in meiner FritzBox das "virtuelle" Telefon eingerichtet, und möchte nun eine Automation erstellen.
Meine FritzBox hat die IP: 192.168.200.1, Home-Assistant läuft auf 192.168.200.6 und "hass-ipp" ist der Name des IP-Phones in der Fritz.
[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 9.5 (armv7 / raspberrypi4)
Home Assistant Core: 2023.1.7
Home Assistant Supervisor: 2023.01.1
-----------------------------------------------------------
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 --ip-addr=192.168.200.6'
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri":"sip:**620@fritz.box:5060","message_tts":"Write here your message"}
Converting audio file 'http://192.168.200.6:8123/api/tts_proxy/3bb4cd06a7898fc0a33665f241cb48f2f2a192ac_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:**620@fritz.box:5060'...
This call will be terminated after '50' seconds.
07:30:45.782 os_core_unix.c !pjlib 2.11.1 for POSIX initialized
13:59:01.783 sip_endpoint.c .Creating endpoint instance...
20:27:17.784 pjlib .select() I/O Queue created (0xb67100c8)
20:27:17.784 sip_endpoint.c .Module "mod-msg-print" registered
20:27:17.784 sip_transport. .Transport manager created.
20:27:17.784 pjsua_core.c .PJSUA state changed: NULL --> CREATED
03:10:29.811 pjsua_core.c .pjsua version 2.11.1 for Linux-5.15.84/armv7l initialized
07:56:37.819 pjsua_app.c .Turning sound device -99 -99 ON
07:56:37.819 main.c Ready: Success
15:17:57.824 pjsua_app.c .......Call 0 state changed to CALLING
Account list:
[ 0] <sip:192.168.200.6:5060>: does not register
Online status: Online
*[ 1] sip:hass-ipp@fritz.box:5060: does not register
Online status: Online
Buddy list:
[ 1] <?> sip:**620@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:**620@fritz.box:5060 [CALLING]