1 I will assume that you have a Fritz.box router with VoIP functionality. Your LAN IP CLASS is 192.168.178.xx
2 you don’t need to specify sip_server_uri options. You can safely remove it from addon config
3 in your configuration.yaml or in you automations.yaml you need to invoke this addon to place a call. So you’ll have something like:
so i have a Script but when save come error:
Message malformed: Integration not found.
please read HomeAssistant documentation and try to make a simple automation before try to use more complex thing.
I don’t use GUI script generator so I cannot help you with this:
WOW its working, thanks
alias: Anruf
sequence:
So what is the best way to call five peaple…my Solution…(at the moment with two)
alias: Anruf call
description: ‘’
trigger:
alias: Anruf Autom.
sequence:
thanks
You can make 5 service call… Or you can use a Virtual pbx like pbxes.com to make an extension that will make ringing all sip client you like…
Hi,
I have successfully made the call. However, the error log is still recorded in the log and the call reception time is quite slow, I don’t know the cause. I appreciate any help (I use GRANDSTREAM UCM6202)
You have 1 active call
Current call id=0 to sip:[email protected] [CALLING]
>>> 11:46:09.457 pjsua_app.c .Turning sound device -99 -99 OFF
**11:46:39.919 tcpc0x7fe4a9052508 TCP connect() error: [code=120110]: Operation timed out**
**11:46:39.919 tsx0x7fe4a90686b8 Temporary failure in sending Request msg INVITE/cseq=2804 (tdta0x7fe4a9061a88), will try next server: Operation timed out**
**11:46:39.919 pjsua_app.c SIP TCP transport is disconnected from xxx.xxx.xxx.xxx:5060: Operation timed out [status=120110]**
11:46:40.241 pjsua_app.c ........Turning sound device -99 -99 ON
11:46:40.241 pjsua_app.c .....Call 0 state changed to EARLY (180 Ringing)
11:46:43.184 pjsua_app.c .....Call 0 state changed to CONNECTING
11:46:43.185 pjsua_app.c .....Call 0 state changed to CONFIRMED
11:46:45.296 pjsua_app.c ......Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
11:46:45.296 pjsua_app_common.c ......
[DISCONNCTD] To: sip:[email protected];tag=12ad81cb-4556-4dd2-9aa0-80f1a4b7c3bd
Call time: 00h:00m:02s, 1st res in 31785 ms, conn in 34729ms
#0 audio PCMU @8kHz, sendrecv, peer=xxx.xxx.xxx.xxx:18952
SRTP status: Not active Crypto-suite:
RX pt=0, last update:00h:00m:01.838s ago
total 103pkt 16.4KB (20.6KB +IP hdr) @avg=63.5Kbps/79.4Kbps
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 0.709 1.750 1.250 0.362
TX pt=0, ptime=20, last update:never
total 90pkt 14.4KB (18.0KB +IP hdr) @avg=55.5Kbps/69.4Kbps
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
What does this mean? “quite slow” is not a technical term.
and there isn’t any info about your system config… Router, IP Class, addon startup log ecc
try, I see a TCP error
pjsua_custom_options: '--no-tcp'
Has anyone managed to use this in a plain docker install, no hass OS or hassio or whatever, just home-assistant in docker.
I built the image but it is looking for supervisor etc. I was just hoping to spin up a docker container in the usual way.
Add-ons uses Supervisor to get commands and give output. Basically it’s a bash script that call an executable.
You can edit the “run” bash script to obtain parameters from command line and place the desired call…
i used this a lot when i had google translate, but now i use nabu casa tts and i need to set the language and gender in my calls- but how to do this in this addon?. i dont have any config in configuration.yaml when i use nabu casa. i set the tts in the addon config and it works but i need to set the gender and language.
Set nabucasa tts as you need and then
The thing is that I want the sip voice to be English and male. But for other automations I use danish and female.
You can Play an MP3 audio file…
If you configure your hassio tts to synthetize in danish female voice you can’t get tts in other languages…
Or you can use another tts platform configured in different way…
add
platform: cloud
to use NABU CASA TTS Service
Thank you very much sdesalve for your really handy addon and all the hard work, you’ve put into creating this. It’s working perfectly with my Fritzbox and is currently used in Fire Alert-Automations, as well as in a self created “Panic Mode”.
Is there any way in showing the CLIP-number to the callers, when i am calling someone through the addon? The number seems to be always supressed, even though i set up the FritzBox to generally showing the numbers for outbound calls. When using the SIP-Number with other calling tools, the number will be shown. Is it a configuration thing in the addon itself, or is it the wrong place to look into in finding it out?
Thanks in advance and stay safe.
Chris
Would you try to set
caller_id_uri: 'sip:[email protected]'
With your fritzbox line phone number?
eg:
caller_id_uri: 'sip:+393331234567'
try also without country code or without leading +
Let me know if this solve your issue
Thanks for the fast reply.
I just tried out all of the constellations, but of no luck. When used with a trailing plus the addon refuses to start at all based on the following error: “pjsua_app_config.c Error: invalid SIP URL ‘sip:+49301234567’ in local id argument”
All the other number variants were used by the addon and with a test call through the dev tools according to the protocol of the addon it was in the state of “CALLING”, but on fritzbox-side there were no sip-registrations active, or any call attempts visible.
This time it accepted the plus in the id, but nevertheless no calls went through to my phone.
I again tried out all the ways possible how to write my number, but no combination worked in the end.
Thanks for your help so far sdesalve. I really appreciate your support.
I actually would be intrigued to know, if i’m the only one here with a surpressed caller id (in combination with a fritzbox).