That’s working fine, it always says that.
This isn’t a “problem”, the instruction to point your browser to the Asterisk SIP HTTPS server is for importing the certificate into your browser if you’re using non-valid, incomplete and/or self signed certificate. This is because you’re going to use SIP over a (secure) websocket with the SIP-HASS stuff which only works over TLS (as explained in detail some posts above). The upgrade required message is there because you’re sending a regular HTTP GET to this websocket, again, this is only for importing the certificate.
As already asked by TECHFox
Does it work when you use the local IP and accept the certificate manually?
If it does you probably trying to use SIP-HASS in an internal network with an external HA (& Asterisk) endpoint (Could be anything like DUCKDNS or other stuff to publish your HA instance to the Internet). If so you need NAT Reflection configured on your router, which in most cases isn’t supported by ISP modem/routers or low-end consumer models. This should be done for both SIP as RTP traffic.
Another way is using a split-dns setup with both internal (LAN) and external (Internet) DNS zones, so you can resolve your SIP server with local addresses.
did it all. And the integration worked to the server. But there are no persons
reinstalled the integration again and here are the logs
Logger: homeassistant.util.async_
Source: util/async_.py:150
First occurred: 22:14:39 (2 occurrences)
Last logged: 22:16:54
Detected blocking call to sleep inside the event loop. This is causing stability issues. Please report issue to the custom component author for asterisk doing blocking calls at custom_components/asterisk/__init__.py, line 101: sleep(5)
I found my problem. I run Nginx. It saved the SSL certificates under folders. I changed my configuration to below and now the certificates work.
OK, so I am using the Addon and integration. Everything is configured and no errors. I am now trying to call from a softphone to the card or the card to the softphone. I can call both ways, but the other device doesn’t seem to ring. I am getting the following log. Do I have a Codec issue?
[May 11 16:01:41] DEBUG[907]: res_pjsip_session.c:4180 session_on_tsx_state: 111 TSX State: Proceeding Inv State: INCOMING
[May 11 16:01:41] DEBUG[907]: res_pjsip_session.c:4184 session_on_tsx_state: Topology: Pending: (null topology) Active: (null topology)
[May 11 16:01:41] DEBUG[907]: res_pjsip_session.c:4189 session_on_tsx_state:
[May 11 16:01:41] DEBUG[907]: res_pjsip_session.c:769 handle_incoming_sdp: 111: Media count: 1
[May 11 16:01:41] DEBUG[907]: res_pjsip_session.c:495 ast_sip_session_media_state_add: 111 Adding position 0
[May 11 16:01:41] DEBUG[907]: res_pjsip_session.c:541 ast_sip_session_media_state_add: Creating new media session
[May 11 16:01:41] DEBUG[907]: res_pjsip_session.c:584 ast_sip_session_media_state_add: Setting media session as default for audio
[May 11 16:01:41] DEBUG[907]: res_pjsip_session.c:589 ast_sip_session_media_state_add: Done
[May 11 16:01:41] DEBUG[907]: res_pjsip_sdp_rtp.c:1504 negotiate_incoming_sdp_stream: 111
[May 11 16:01:41] DEBUG[907]: res_pjsip_sdp_rtp.c:1508 negotiate_incoming_sdp_stream: Endpoint has no codecs for media type 'audio', declining stream
[May 11 16:01:41] DEBUG[907]: res_pjsip_sdp_rtp.c:1510 negotiate_incoming_sdp_stream: Endpoint has no codecs
[May 11 16:01:41] DEBUG[907]: res_pjsip_session.c:960 handle_incoming_sdp: 111: Handled? no
[May 11 16:01:41] DEBUG[907]: res_pjsip_session.c:4514 handle_outgoing_response: 111: Method is INVITE, Response is 488 Not Acceptable Here
Finally, some progress is achieved 
All my persons are identified if “main” integration version is installed. It did not work for either “0.4.x” or “0.6.x” versions. Though “main” integration version creates not binary_senor, but two sensors for each person. Sensor.state and sensor.callee. How that could be fixed? Thanks for your efforts.
Stay away from integration, it’s not finished yet… Start with the sip card and addon
thank. But can you describe in more detail how to create a binary_senor for each person without integration?
to make call sensors? or why do you want it?
well, on the map, which entities to show ?
what map ?
- person: person.papa # Your first person entity here
name: Person1
extension: '101'
secret: '###' # Set the auto_generated_secret you set in the add-on here!
icon: mdi:monitor
entity: ``` ---what to post here ?nothing, remove it
tried. No response to call. And there are no mistakes
check your logs?
There are no errors in the log. The error is only in the map itself person not cofigured! There is no extension configured for this person
But all the personas I really created
That error means there is no extension configured with the person you are currently logged in as. To use the card, the logged in person needs to have a extension configured, so the card knows which extension to use.
So make sure your person has a user that can log in, and use that one.
Seems like it. Make sure the codec the softphone sends is configured in the endpoint.