This thread should be split, the SIP thingy was a hijack, the OP was all about debugging IAX2 trunks.
However, if anyone has a ubiquitous and guaranteed NAT traversal method for every scenario of every implementation of NAT on every router from a DLink to a Cisco including the notorious SonicWalls and every Vendor supplied Docsis/DSL boxes out there, never mind double NATting , then posting it would make her an instant hero. I doubt that will happen here or elsewhere any time soon :-). I believe that IAX2, perhaps on another port if necessary in some countries/ISP’s, will largely work, that was it’s initial design concept. So to paraphrase Tolkien
One Port to Rule them All.
This is my last word here on this thread about SIP
When you are Qualifying trunks/endpoints with Chan_SIP, PJSIP and even IAX2 Asterisk expects replies to the OPTIONS Keep-Alives it sends to the peer. It generally expects this reply in 2000ms (2 seconds) and if it does not get this reply it will “Re-Transmit” the request again. It re-transmit like this for at least seven-ish attempts after which it will mark the peer UNREACHABLE, because as far as Asterisk is concerned, it can’t reach it. It will attempt this process again to try and make the peer REACHABLE/AVAILABLE and if it can’t it will clear the registration from memory. Now if this static peer, like in your case, there is no registration to drop from memory, it will just continue to be UNREACHABLE.
The cause of your issue is either NAT or the network. Since this is over a VPN, there is good change it is a combination of both or at least something with your routes over the VPN. You need to look at your VPN, the routes and any NAT you have setup for the VPN.
Let’s be clear here, IAX/IAX2 was never a magical NAT fix. The benefit of IAX/IAX2 for NAT was the fact it was ONE port, 4569, for the signaling AND media. SIP (Chan_SIP/PJSIP) requires a port (like 5060) for the signaling and then a wide range of ports for the media (generally 10,000 to 20,000) because SIP requires two ports for media (RTP/RCTP) per call, so if a router had poor NAT handling or was just crap all around it made SIP a problem because the audio ports would also be the problem and you ended up with no audio/one-way audio, etc.
But that issue, just like the need for g729 back then, is now really a non-issue. Routers these days have better NAT handling all around along with improvements specifically for SIP. Why? Because SIP is the standard that ALL providers/carriers use. Providers that provide IAX2 trunks are specifically running Asterisk servers while the rest of the carriers/providers will probably only know what IAX is if anyone there has ventured into Asterisk territory.
So again, your issue is a NAT/Network issue. I would be looking into your VPN connection between the two sites.
I’m sorry guys but due to an altercation between two users that happened privately in regards to this thread over who is smarter than who and who’s ego is bigger than who’s I am closing the thread.
For some reason the topic at hand and the discussion that came about from it caused certain users to get extremely heated. This thread is being closed to prevent these issues from “spilling” over.