SIP client card, as intercom

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343

Maybe you also need to open port 5060? I suggest trying that.

344

I redirected to this path / etc / asterisk and it still does not read these sound files

345

Unfortunately it didnā€™t work. I can call between the extensions, as well as answer. But if Iā€™m on a non-local network the audio doesnā€™t work. I did the following: port 5060 to 5060 protocol UDP, port 8089 to 8089 protocol all and ports 10000-20000 to 10000-20000 protocol UDP

346

You said 10000-2000 to 10000-20000. This should be 10000 to 20000. And I would set all protocols to all just to be sure.

347

Weird. I will try to do it myself this week. I will let you know.

1 Like
348

I changed the protocol to all on all port forwards, but it didnā€™t work. I found that despite the configuration on my router, when I consult this site Open Port Check Tool - Test Port Forwarding on Your Router ports 5060 and 10000-20000 are still closed. So the problem must be with my network or router.

349

Try to find some sip settings on your router and deactivate them .which h router do U have ?

350

tp link deco m5

351

It has a configuration called SIP ALG and the description says that when activated, clients can communicate with SIP servers (Session initiation protocol) via NAT. This setting is currently enabled. Should I disable it?

352

SIP ALG feature on routers usually causes problems on SIP connections and it is best practise to disable it

353

I disabled my routerā€™s sip setting but it still has no audio.

354

try this:
enable the HA on DMZ IP, to check if it is a port forwarding issue and then disable it again

355

I confirm .disable it ā€¦what day the logs on the browser?I had to use stun server in my freepbx to get it working

356

Is anybody getting the following message. I just upgraded to 6.0, but even if I downgrade, I still get the message.

image
image469Ɨ552 18.3 KB

357

sorry I have downgraded to 4.2 and every thing works. So I think it could be something introduced into 5.0 and 6.0

358

Everything works fine now, and extensions can call each other. but
There are multiple asterisk extensions logged in homeassistant, can broadcast function be added to HA-SIP? For example, when there are three screens.

359

I was finally able to get this working with the companion app. I have Home Assistant behind a reverse proxy (via the Nginx Proxy Manager add-on) using a Cloudflare Origin Certificate. Hereā€™s a quick summary of what I did.

  1. Add a DNS A record on the Cloudflare dashboard. Iā€™m accessing my Home Assistant instance through my root domain, so I setup a sub domain record.

  2. Set up a proxy host in Nginx Proxy Manager. Use the address from step 1 for the domain name and forward port 8089.

  3. Set up port forwarding on your router. I got the list of ports from this guide:

    • UDP Port 5060
    • TCP Port 8089
    • UDP Ports 10000-20000

  4. Update the Asterisk Add-on to use the Cloudflare cert. I used Samba to place it in the SSL directory, it will not load it from the config location.

  5. Set up the card to use the new server path. For example,

type: custom:sipjs-card
server: sip.jojeda654.xyz
port: '443'
2 Likes
360

6.0 is a test version. The error will be fixed in the next version. :slight_smile:

361

My intercom calls extension 200. I would like that every time extension 200 rings, extensions 100 and 101 also ring simultaneously until the first one picks up.

My extensions.conf file looks like this:

[general]
static=yes

[default]
include => parkedcalls
;
exten => 444,1,Park(,s)
;
exten => 555,1,ParkedCall(default,701)
;
exten => _X!,1,Dial(SIP/${EXTEN})

[parkedcallstimeout]
exten => s,1,Hangup()

exten => 200,1,Dial(SIP/200&SIP/100&SIP/101)

But it does not work. Extension 200 rings and nothing happens like extensions 100 and 101

362

Guess you need to place it in [default], above exten => _X!,1,Dial(SIP/${EXTEN}).