All calls fails after 31 seconds

Depends on your firewall/router. Look for a section entitled port forwarding

Are the phones on the same local network as the PBX?

No. Phones in
Asterisk in

So they are local as in not going over the Internet. If you cant make extension to extension calls with audio, you dont have things configured right on the pbx.

Describe how that is setup.

I can do calls with audio. Extension to extension. Extension to real number. Real number to my telephones.

But do all your calls drop after 31 seconds?

I did a quick Google search and found these similar, but resolved issues. Did you have a chance to look at these:

Yes, all calls drops

@comtech That is a shotgun approach to this. Did you read any of those to determine if they were duplicate or even relevant to this issue at all?

@necto_random Alright so you probably have your PBX configured improperly for the network setup. If all the calls, including those on the local network, are dropping after 32 seconds the RTP/media IPs details are either not correct or something in the local network is messing with the traffic.

What do you have in your Asterisk SIP Settings? All three tabs: General, Chan_SIP, Chan_PJSIP

Now server is down) I’m at home.
I write about this at the mondey.

When i call all lines is busy.
Call is in freepbx queue, but it drop at 32 second too.

Two questions:

  1. Do you have added under Settings > SIP Settings > Local Networks?
  2. Did you forward ports 10000-20000 (or whatever your RTP port rage is) to the PBX?

Now only incoming calls falls. Server freepbx was only rebooted.

Why don’t you follow the instructions or answer our questions here?

i do it, i add advanced in pjsip, added local interfaces and ports fnd it wark correctly

You’re still not answering our questions about your setup and what is actually configured. You’re having network/NAT issues and to be honest, you’re using a Mikrotik so that means the only way this is screwed up is because the Mikrotik has been setup wrong. Those things Just Work ™ with SIP setups and I’ve never seen them have issues like this unless they were messed with and rules are screwed up.

You need to actually answer our questions and provide us with the details we asked for.

now all works correctly. i solv problem

And what did you do?

I addad ports and networks for pjsip

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