Im still trying to work my way around this, so Id appreciate any help anyone can offer!
My Asterisk/FreePBX server is on a DSL connection at home, behind my
router but on a Public IP address. I did a little testing with the
firewall disabled so I dont think that is contributing to my problems. I
have a couple of Sipgate trunks defined and also a SPA-3102 connects my
PSTN line to Asterisk and my home phone as an Extension.
I set up an IVR as a inbound route from one of the Sipgate trunks to basically be able to dial any extension.
When using this at home, it works great. Running Eyebeam on any PC
and the calls works fine when calling into the SIPgate number from my
mobile phone. This works equally well when the Eyebeam extension is used
from my work and Im VPNd onto my router at home.
However, when Im not VPNd into my own network, Eyebeam rings, I click
answer and the display says Call Established. However the phone that Im
ringing from continues to give me the ringing tone.
Ive looked at the Asterisk CLI and it looks like the Answered message
is not received. Im not entirely sure how to turn on more logging.
Im wndering whether this could be down to NAT issues? The Extensions are set up with NAT=YES on the FreePBX webscreens.
Being on your home network and using a VPN both accomplish the same thing and put your Eyebeam on the same local network as your Asterisk. Could very well be a NAT issue as some routers (especially consumer routers) are better than other at handling SIP. If you have SIP ALG or SIP helper turned on in the router turn it off. I would also look closely at the RTP/audio streams; the Asterisk should have tcpdump installed.
And let me correct my self i configure freepbx on server which have public ip. i installed freepbx with distro. I configure extension in eyebeam and configure trunk is resister is given below:-
From you describe it sounds like while the PBX has a public IP, it is still behind a firewall and you are doing some kind of port forwarding or DMZ; this would explain why your trunk uses NAT. I would take Mark’s advice and make sure the all the necessary SIP and RTP ports are properly forwarded to the PBX in your firewall.
Assuming the PBX is NOT behind a firewall, the NAT on the trunk is unnecessary and is not really doing anything. Do you have two NIC connections, one for the public IP and another for the local LAN? How do your devices connect to the PBX on the local LAN? It would seem a waste of resources to have them go out to the internet just to come back into the local LAN and connect.
Additional logging can be turned on in the CLI with the SIP debug command, be sure to turn it back off when you are done it fills the log file quickly. The trouble bears many of the earmarks of a NAT issue but the setup in general seems unnecessarily complicated considering the PBX and the devices are physically located together.