SIP client card, as intercom

Yeah I mean registered. Trying to find a way to keep it registered. It should work fine if the card is a button that I can keep on my main view. The tablets default back to that view
Every 10 minutes of inactivity. The other option would be to make some sort of footer that has it in it. The interface doesn’t need to be large (I think it needs to be visible though) to register. I wonder if there is a way to register in the background.

Why not just hide the card on your main page then? Problem solved? On incoming call, just unhide it ?

Registration will always be an issue with the websocket behind, even if on mainpage… If browser/app/… closes, or you browse away, it gets unregistered… A button will not help

I guess I just don’t get the logic how to unhide it on a call unless it’s registered? I could use browser mod to popup the card on app the tablets if someone opens it. Is that may your thinking?

I don’t know your situation, you have some android tablets on the wall? They are probably on some kind of default main view… Well, just add the card on the main view, but hidden?

On incoming call, unhide the card… isn’t that your goal?

But anyway, it will always be difficult and looking for workarounds, the issue is the wss connection, it’s much easier to just use parked call, no need for registered state

You can also do instead of dial, a redial in the dialplan

If the card is unregistered, then it tries calling again and again and again untill you open the view where the card is active, it gets registered and you can see the call coming in

Guys I keep posting and posting here and on Discord.

I just can’t get it working, I already tried the tutorial four times and I always end up nowhere…

I know it’s quite complicated, but come on it’s not the third secret of Fatima and we should finally be ALL able to enjoy this unique integration.

Can we (I dont think I am the only one right?) please, please, please, please, please, please, get a 100% working step-by-step guide?

I don’t want to sound harsh as I know the author has put BIG effort into doing docs and I am grateful for that, but hearing also other peeps (on Facebook) they are all in the same boat. :sob:

I personally have an Amcrest AD410 so I know it is SIP compliant.

We need more info, what have you done? what doesnt work? what are the erros, logs … how is your setup ? The tutorial is just a guide … there is no step by step guide … everyone has their own custom setup… You need to be familiar with SIP, maybe its best to follow some basic asterisk guides to learn about SIP , then try again this integration … Enough wikis and guides online to learn basic SIP protocol… start with softphones on your mobile, setup 2 extensions, try calling to each other …

Well Fabio (btw I am Italian too) I know skilled (Asterisk/HA,JSON, etc.) users that ended up like me so I am here basically to beg someone to get this working for everybody.

I know setups and hardware are different for every user, but this is the first and only project where I got stuck after MANY attempts and MANY readings.

Again I am not complaining I just think this integration is the one and only MUST for every video doorbell HA user and I can’t have HA without it! :smiley:

PS: by the way Fabio if you would be interested into joining an Italian Facebook Home Assistant group I think you could help MANY of us lol!

OK :slight_smile:
But again, to start helping, povide us more info, logs, troubleshooting, screenshots, things you tried, whats the actual issue …

so where is the log? you stopped answering?

image

My guess you are using the WAN IP to connect, make sure you use the domain name instead, and make sure your connection is then secure with a VALID ssl , since you are using the IP instead, probably your SSL doesnt match , and you get connection issues then

Easy man I am at work right now so I can post only now & then! :joy:

Let me collect everything again once I get home and I will tag you when posting. :+1:t5:

no need to tag :slight_smile:
But that last message was from 07 April ? Thats not “now & then” :slight_smile:

Mate it looks like you are taking it personally for some unknown reason, but you are getting it wrong. :wink:

As I said I am not here to flame or anything.

I just wanted to point out that many of us are not able to get everything working and the mistake is probably on our side.

That being said I say let’s go back in topic. :slight_smile:

1 Like

Hi,
I’m trying to add a Dahua VTO (ext. 8001) to call my wall-mount iPad (ext. 103) using the Asterisk addon, integration and the card. If the iPad’s screen is on, I call call its extension and it works fine with camera and two-way audio as it should. However, the moment the iPad goes to sleep the card gets unregistered and the number is unavailable. Unfortunately I can’t even wake the iPad with an automation of call status of 8001, since it does not change. So I looked up the docs site and found parking. I set the VTO to call 444 and then I go to the iPad or my phone (101) to call 555. It starts a call but nothing is going through, as after a while the VTO drops the connection for “not responding”.
According to the logs the connection is made though, but I can’t hear anything on either end and I don’t see the camera feed. What should I do?

Here is the log: [May 24 21:51:13] == Spawn extension (default, 555, 1) exited non-zero on 'PJS - Pastebin.com

Thanks

Did you manage to get it working? If not, please use discord. I rarely check here.

I’m not on Facebook, but I suggest posting the discord link there. People can help them there.

When the outdoor doorbell calls, both indoor tablets ring simultaneously (100and101), and either can be connected. Is this function ‘parking’ or ‘Conference’

Has nothing todo with parking or conference, its your dialplan, probably you are just doing a dialgroup command, that means, all extensions will ring that you added in the group , or maybe its a fucntion of your doorbell ?

Please provide an example syntax in extensions. conf

An example of what , dial group?

Google => Asterisk dialgroup
Then open first hit

https://wiki.asterisk.org/wiki/display/AST/Function_DIALGROUP