IP authentication between Asterisk 1.4 and FreePBX 2.6

FreePBX Installation / Upgrade
1

Ok… I’m trying to figure this out… I got Asterisk 1.4.26 and FreePBX 2.6 (with Asterisk 1.6.2.4). What I’m trying to do is to allow FreePBX pass all outbound calls to Asterisk for routing and also all inbound calls should be coming from Asterisk.

After taking a look at a different posts, I found a couple of samples (broadband.com sample was probably the most helpful). However… While I can make an outbound call from FreePBX through Asterisk, I can not get the incoming. I keep getting “603 Decline error” coming from FreePBX. I tried to enable sip debug on the peer on FreePBX but I am not seeing anything coming in…

Another thing that someone complicates the issue - FreePBX is behind a security appliance but I did setup a rule for inbound and outbound traffic. I also had nat=yes on both - FreePBX and Asterisk

Anybody who can shade some light on what settings need to be used on both sides?

Any help is much appreciated.

2

This is a confusing post. What you have is two Asterisk systems… 1.4.26 and 1.6.2.4, the later you manage the dialplan with FreePBX.

How is the dialplan on the 1.4.26 box managed ? “Security Appliance” is a bit vague as well.

Try googling “sip trunking” and/or “iax trunking” and see if that gets you anywhere.

3

Correct… 1.6.2.4 managed using freePBX. 1.4.26 - manually… by hand… Security Appliance is Astaro, box is capable of filtering not only incoming but also an outgoing traffic. Port forwarding is setup and VoIP protocols (covering port 5060 and RTP range 10000-20000) are configured to be allowed.

FreePBX is behind Astaro, pure Asterisk - public IP. I can make call from FreePBX to Asterisk but not other way around. Most of the samples I found on google have to do with register => while I’m trying to do a pure IP Authentication/Hand off. So what I’m trying to get an idea of is:

  1. How many entries on a pure Asterisk do I need to make to make this work and what types of connections? Peer? Friend? user?
  2. Same thing but on FreePBX.

Again… the goal is to avoid using “register” as it does create overhead that I’m trying to avoid.