And obliviously please check that you don’t have any firewall on your NUC or on your LAN that drops SIP call packets…
Is there an option that avoid LAN outgoing SIP packets on Fritz!Box router for example…
Initially I waited couple of hours and now its 24 hours and counting so should not be that. And I can hear the intended test TTS audio from the link as well.
I have my NUC through an access point so first removed that just in case and connected directly to my 4G modem (did not help). Then I disabled the firewall from the 4G modem and checked that it had an option called SIG ALG (which was already turned on) but this did not help either. And the NUC is currently just pure HASSOS so no firewalls on that I assume.
Just to confirm the last point I also had second instance of HA available on Raspbi which I installed the addon as well but that did not work either (basically same log output with just the host different).
I could not eliminate my service provider from the picture (though the modem is theirs and it has this SIP option so I assumed they support it as well). I’ll try to get the Raspbi to work on my phone’s wifi hotspot to rule this out still.
Could the FreeVoipDeal be the issue? I’ll check if I can try another provide without putting more money in. Also I did not understand the “I’ve noticed that you don’t have an https URL for your HomeAssistant” -> can this be the issue somehow?
I use FreeVoipDeal also and it works very well.
Obviously you have some money on your account, it’s true?
I don’t think can be this your problem. But if I’ve understood correctly, you use a 4G connection and your IP should be behind a NAT so I don’t think you can use DuckDNS addons
Yes, put some credit there and tested the calling via the site itself so that seems ok.
So the 4G might be the issue (Finnish DNA). I checked the service providers pages and saw some forum posts about PS4 related public IP/NAT config. Ive not set any port forwarding as an example but understood that it could be the route.
Ill try to get duckDNS addon next. Thanks again for your support!
hello sdesalve,
I just testet your addon. it works fine for internal and external calls on my fritzbox 7490.
just one question: what would be the correct URL to play local files? … http://:8123/config/mp3/xxx.mp3 does not work
thanks - mat
when I enter the URL http://my-ha:8123/config/mp3/xxx.mp3 in the browser (for testing) i end up in the lovelace (GUI).
in the log i see this:
[Info] Listening for messages via stdin service call…
[Info] Received messages {“call_sip_uri”: “sip:**[email protected]”, “audio_file_url”: “http://my-ha:8123/config/mp3/ColdMetal.mp3”, “max_call_time”: 30}
Converting audio file ‘http://my-ha:8123/config/mp3/ColdMetal.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.
Specify Text-to-speech platform to use. Default value if this option is not specified is google_translate. For a list of available TTS integration please see Hassio integrations
https://www.obitalk.com/info/googlevoice Basically lets you use a free google voice account to make voip calls. Im not 100% sure but i think you can setup the obihai to do a sip call.
1 it’s normal. Pjsua command is interactive and it will run until max_call_time
2 are you sure? In 1 question you stated that max_call_time was respected. Please recheck config
I am trying to use this add-on. I a having trouble running a call.
I use messagenet. I got the apk running on my android phone.
The setup is OK as defined here.
The log shows the following and I have no idea what to do.
I do not get any call on my mobile.
[Info] Listening for messages via stdin service call...
[Info] Received messages {"call_sip_uri": "sip:[email protected]", "message_tts": "Alarm Armed"}
Converting audio file 'https://myHAservername:8123/api/tts_proxy/d3cd3c303bb602ef28458c0e4c1c9bbf01d17dbc_en_-_google_translate.mp3'...
Audio succesfully converted...
Starting SIP Client and calling 'sip:[email protected]'...
This call will be terminated after '50' seconds.
21:27:29.751 os_core_unix.c !pjlib 2.9 for POSIX initialized
21:27:29.752 sip_endpoint.c .Creating endpoint instance...
21:27:29.752 pjlib .select() I/O Queue created (0x178de78)
21:27:29.752 sip_endpoint.c .Module "mod-msg-print" registered
21:27:29.752 sip_transport. .Transport manager created.
21:27:29.752 pjsua_core.c .PJSUA state changed: NULL --> CREATED
21:27:29.768 pjsua_core.c .pjsua version 2.9 for Linux-4.19.127/armv7l initialized
21:27:29.771 pjsua_app.c .Turning sound device -99 -99 ON
21:27:29.771 main.c Ready: Success
21:27:29.835 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]
