SIP client card, as intercom

Share your Projects! Custom Integrations
303

You make the link to a person with the sip card

304

Thank you for your help!

Now i think i configured everything correct… but its saying connecting offline.

type: custom:sipjs-card
server: 192.168.my.ip
port: '8089'
custom:
  - name: Custom1
    number: '123'
    icon: mdi:phone-classic
dtmfs:
  - name: dtmf1
    signal: '1'
    icon: mdi:cow
ringtone: ''
ringbacktone: ''
extensions:
  - person: person.me
    name: me
    extension: '101'
    secret: ''
    icon: ''
    entity: person.me
  - person: person.wife
    name: wife
    extension: '102'
    secret: ''
    icon: ''
    entity: person.wife
  - person: person.test_user
    name: Test
    extension: '103'
    secret: ''
    icon: ''
    entity: person.test_user
video: false
305

Yes, probably, put in a hostname instead of an IP, otherwise you have an invalid SSL certificate

306

i tried localhost but its not working

307

no, put in your domain name, like for example your nabucasa or duckdns or whatever :slight_smile:
then open port 8089

it uses the SSL certificates files by default

308

Ok now its connected but when i press call it opens a call popup but doesn’t do anything on the device logged into the extension im calling. what di i need to do? Again thanks for all your help!!!

309

hmm, not sure, should work :slight_smile:
you can maybe check the logs? please join also discord, lots of users there that can help

310

i am on a vpn wont be home for a while could that be it?

311

no idea :slight_smile:

312

LOL thanks anyway for all your help so far

313

If someone has an existing PBX that shall be reused instead of adding another asterisk PBX, then maybe this is a starting point for you:

For all who do not know the “Fritzbox”: It is Wifi-Router with basic SIP-PBX that is popular in Germany.

But instead of the Fritzbox any SIP server could be used.

The main idea of this docker container is that the well-known SIP proxy kamailio is configured in a way that it just acts as a websocket endpoint and forwards any SIP message to the real PBX which could be any SIP PBX.
As SIP-over-Websockets also requires the use of WebRTC, the docker container also includes a corresponding rtpengine which acts as a media bridge between webrtc and the normal RTP world.

kamailio+rtpengine make this possible: SIP-over-websocket+WebRTC <-> SIP (UDP, TCP, TLS)+(S)RTP.

The REGISTER message is NOT processed by kamailio but forwarded to the real registrar which is gathered from the SIP uri.

Just for your info… :wink:

1 Like
314

@TECHFox
Troubleshooting question:
How do I access the asterisk CLI of the addon to manually debug my config?
Normally I would issue something like asterisk -rvvvvdddd to enable the CLI with debug logs.

315

Just enter the docker container with the ssh addon

316

You can also enable the debug logs with the log_level option. You can see the different log levels here.

317

@meni123 there is a new beta version (0.5.10) that includes the callee sensor which displays the number it’s calling. So you can for example trigger automations by calling a extension.

More sensors and services are coming. :slight_smile:

1 Like
318

Surprise, it’s perfect, i’m running to check it out (:smile:
you are great!

319

@pergola.fabio
I have set up a private domain for my HA, I have an SSL certificate
Which port do I need to open for an asterisk to work properly, my asterisk is not external but a plugin made by @TECHFox

320

@TECHFox Thank you, the sensor is for outgoing calls, an incoming call sensor would be more helpful,

321

Depends if you want to use softphones or the sip card?
RTP os needed for audio…

5060 for softphones

322

@pergola.fabio
use both the softphones and the sip card,
What should I open?