@JaredBusch said in FreePBX : Skyetel inbound call "Rejecting unknown SIP connection ...":
@BraswellJay said in FreePBX : Skyetel inbound call "Rejecting unknown SIP connection ...":
I'm thinking this is firewall related. I forwarded ports 10000-20000 on inbound connections to the freePBX server and this allowed for audio from external caller to one of our extensions, but still no audio from extension to external caller.
Outbound calls are ok over this trunk. Two way audio is good.
First you need to understand call flow.
When you make a call the PBX is reaching out to the SIP provider on WTF ever port you have setup. The PBX is the one in charge of setting up the ports for the audio. The PBX is the one sending information constantly about the call. There is no need to worry about port forwarding because the ports are automagically opened by the NAT setup like any other outbound connection.
When you receive a call, the provider side is doing all of that. Nothing is initiated from the PBX side. SO you have to ensure the inbound information can get through to where it needs to, aka port forwarding and firewalls.
So that means you need to make sure that you are forwarding all the right things. Additionally, you should only allowing this traffic from the @Skyetel IP blocks, but that is not important immediately as you have things not working right. Fix that first, then secure it properly.
I had limited the incoming SIP traffic to just the Skyetel IP blocks (albeit I had the poorly implemented port forwarding going on as discussed in the other posts in this thread) but I had left the forwarding of the udp/10000-20000 as from any outside IP. I thought this was necessary for this block because Skyetel mentions on their site that their system architecture is such that their servers are not involved in the audio path and that audio comes directly from the PSTN.
Maybe that's bad thinking and I need to lock down udp/10000-20000 to just their IP's as well. I'll experiment with that tomorrow.
So your PBX is designed to use a PJSIP based trunk on port 5160. This means INBOUND SIP TRAFFIC that you want to use the pjsip channel driver needs to come in on port 5160. This doesn't mean dick for outbound. You are behind NAT, your outbound connection might come from any port on your router.
Your PBX is also designed to use the port range 10000-20000 for RTP (the audio) traffic.
You need to port forward udp/5160 to your PBX.
You need to port forward udp/10000-20000 to your PBX.
This is the extent of your router changes. On the assumption that a port forward setup obviously includes the router firewall also.
On your FreePBX system, you need to have the @Skyetel network IP blocks whitelisted in the FreePBX firewall as I mention in post two of my topic on setting up a Skyetel PJSIP trunk.
I had seen this thread that you had done and have made those changes.
Once you have adjusted your router, FreePBX settings, and Skyetel Endpoint settings to all match, I would expect your problems to go away, or at least change.
The 30 second cut off you are experiencing are liekly because things are not talking back and forth to each other on the same ports and so the call is being hung up assuming the other side timed out or dropped offline.