Pjsua Is intended to be executed as interactive program so I’m not able to get a call status feedback or make audio file start playing when call was picked up.
I can’t find such options in pjsua man’s pages or on internet
Hi
I’m relatively new to Home Assistant.
I’m very interested in your VOIP add-on.
I’ve probably a very stupid question. I don’t find your add-on in the HomeAssistant UI (not in official, not in community add-ons) And not in the integrations.
I can download your files but how can I install your add-on on my Raspberry Pi? . Do I need to copy your folder to a folder on my PI? I can do it with Samba. (I don’t know a lot of Linux)
Thanks for your answer.
I was able to add the “add-on” of @sdesalve
I’m new, so you learned me something
Great piece of addon. I have it working calling to my linphone sip account , but when I try to call a local number in my freepbx server it fails with:
pjsua_app.c .....Call 0 is DISCONNECTED [reason=403 (Forbidden)]
any clues or do you need more info?
please post full logs, check also sip auth info in configuration (eg try to place call with microsip softphone with same account)
my mistake. forgot single quotes on the user and password… rookie here…
but max call time seems to be broken. it calls 50 sec before disconnection: aka:
max_call_time: '11'
logs and configurations!
ok…
sip_parameters:
caller_id_uri: 'sip:[email protected]:5060'
realm: '*'
username: xxxxx
password: xxxxxx
max_call_time: '11'
[Info] Received messages {"call_sip_uri": "sip:[email protected]", "message_tts": "Tak for dit tryk. Jeg er på vej til dig."}
Converting audio file 'http://10.0.0.191:8123/api/tts_proxy/359c6b75f667dfbd89e449780443df31b4bb4e3b_da_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '50' seconds.
23:01:53.387 os_core_unix.c !pjlib 2.9 for POSIX initialized
23:01:53.387 sip_endpoint.c .Creating endpoint instance...
23:01:53.387 pjlib .select() I/O Queue created (0x557a266c2c90)
23:01:53.387 sip_endpoint.c .Module "mod-msg-print" registered
23:01:53.387 sip_transport.c .Transport manager created.
23:01:53.387 pjsua_core.c .PJSUA state changed: NULL --> CREATED
23:01:53.395 pjsua_core.c .pjsua version 2.9 for Linux-5.4.86/x86_64 initialized
23:01:53.395 pjsua_app.c .Turning sound device -99 -99 ON
23:01:53.395 main.c Ready: Success
23:01:53.396 pjsua_app.c .......Call 0 state changed to CALLING
>>>>
Account list:
[ 0] <sip:172.30.33.5:5060>: does not register
Online status: Online
[ 1] <sip:172.30.33.5:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:[email protected]:5060: 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]
>>> 23:01:53.494 pjsua_app.c .....Call 0 state changed to EARLY (180 Ringing)
23:01:53.852 pjsua_app.c .....Call 0 state changed to EARLY (180 Ringing)
23:01:55.232 pjsua_app.c .....Call 0 state changed to CONNECTING
23:01:55.233 pjsua_app.c .....Call 0 state changed to CONFIRMED
>>> 23:02:43.383 pjsua_app.c .....Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
23:02:43.383 pjsua_app_common.c .....
[DISCONNCTD] To: sip:[email protected];tag=cf10f916-a2de-43ea-ae8e-08e9ea30cc7e
Call time: 00h:00m:48s, 1st res in 99 ms, conn in 1838ms
#0 audio PCMU @8kHz, sendrecv, peer=-
SRTP status: Not active Crypto-suite:
RX pt=0, last update:00h:00m:02.946s ago
total 0pkt 0B (0B +IP hdr) @avg=0bps/0bps
pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
TX pt=0, ptime=20, last update:never
total 1.7Kpkt 273.2KB (341.6KB +IP hdr) @avg=45.4Kbps/56.7Kbps
pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
RTT msec : 0.000 0.000 0.000 0.000 0.000
23:02:44.382 pjsua_app.c ..Turning sound device -99 -99 OFF
[Info] Call ended...
[Info] Listening for messages via stdin service call...
remove 2 spaces in front of line “max_call_time: ‘11’”
it isn’t a child of sip parameters
1 Like
thank you very much. works fine now!
Sorry for my question. What I did was too stupid to post it again.
Your add-on is working fine now.
I’ll certainly buy you some coffees.
1 Like
Hi
The add-on works fine when I send a 'text to speak tts" message to my mobile.
When I replace the tts message with an audio file, the add-on hangs every time
How did you arrive to play an audio file?
I tried with several MP3 files, in several folders. Using https instead of http. Not sepcifying the :8123 port of my raspberry.(ip=192.168.0.130) Always the same problem.
The Service data
{
“addon”: “89275b70_dss_voip”,
“input”: {
“call_sip_uri”: ‘sip:[email protected]’,
“audio_file_url”: ‘http://192.168.0.130:8123/media/sweep.mp3’
}
}
The log file shows “converting” and than the add-on terminates.
[Info] Listening for messages via stdin service call…
[Info] Received messages {“call_sip_uri”: “sip:[email protected]”, “audio_file_url”: “http://192.168.0.130:8123/share/dss_voip/sweep.mp3”}
Converting audio file ‘http://192.168.0.130:8123/share/dss_voip/sweep.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.
Create a www folder in /config folder
Place on it a file.mp3
URL Will Be http://192.168.0.130:8123/local/file.mp3
Or whatever URL you have configurare as host
Please read hassio docs and how to do basics procedures
Thanks
Now I can play also an .mp3 file.
I was somewhat confused because in Node-Red, I could play on my speakers the .mp3 files which were stored in the /config/media folder. By using: {“media_content_type”: “url”, “media_content_id”: “https://192.168.0.130/media/a_file.mp3”}
I also think that a reboot was also necessary (after creating the www folder and copying the .mp3 files) before the add-on no longer terminated.
By the way: Why the (same) audio file needs to be converted each time? Isn’t this a waste of resources? Is there a way to play an audio file that was already properly converted?
My log file:
[Info] Received messages {“call_sip_uri”: “sip:[email protected]”, “audio_file_url”: “http://192.168.0.130:8123/local/test.mp3”}
Converting audio file ‘http://192.168.0.130:8123/local/test.mp3’…
Audio succesfully converted…
Starting SIP Client and calling ‘sip:[email protected]’…
bashio::log.green "Converting audio file '$URLFILEMP3_VALUE'..."
sox -V3 $SOX_CUSTOM_OPTIONS_INPUT_FILE_VALUE $URLFILEMP3_VALUE $SOX_CUSTOM_OPTIONS_OUTPUT_FILE_VALUE /share/dss_voip/dss_message_tts.wav pad 0.5 1.5 2> /share/dss_voip/dss_sox.log
if [ $? -eq 0 ]; then
bashio::log.green "Audio succesfully converted..."
it is a development choice to allow you and other user to use SOX command line options…
Hi.
I get fatal error in log when starting it. Any ideas? I’m a bit green still.
Hope to get it solved, this Addon would help me a lot.
Log:
[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: 3.5.3
You are running the latest version of this add-on.
System: Home Assistant OS 5.10 (armv7 / raspberrypi3)
Home Assistant Core: 2021.1.5
Home Assistant Supervisor: 2021.01.7
[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
exec: fatal: unable to exec bashio
: No such file or directory
execlineb: usage: execlineb [ -p | -P | -S nmin | -s nmin ] [ -q | -w | -W ] [ -c commandline ] script args
[services.d] done.
exec: fatal: unable to exec bashio
: No such file or directory
execlineb: usage: execlineb [ -p | -P | -S nmin | -s nmin ] [ -q | -w | -W ] [ -c commandline ] script args
exec: fatal: unable to exec bashio
: No such file or directory
execlineb: usage: execlineb [ -p | -P | -S nmin | -s nmin ] [ -q | -w | -W ] [ -c commandline ] script args