Test Environment = FreePBX 18.104.22.168, Asterisk 10.0.0, Centos 6.2. LAN is all static IPs. There is nothing between the phones and the PBX except an Ethernet switch. Phones (end points) are Aastra 6730i’s
PBX status shows all green. sip show peers - shows all phones with correct phone numbers and IP addresses. Under Dyn nothing is showing.
Linux is a very versatile system and I realize things can be different than described following. From the Asterisk 10.0.0 spec -
https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts - Creating SIP Accounts
https://wiki.asterisk.org/wiki/display/AST/Registering+Phones+to+Asterisk - Registering phones to Asterisk
sip.conf is buried deep in /usr/src/-------> etc. and shows no phone detail per the spec. Watching the Aasterisk CLI console on a phone reboots, nothing shows which is again contrary to the spec. The phones themselves throw a 403 error which is listed as hearing the PBX but not liking what they hear and refusing to reply. The phones have no trouble communicating with or taking provisioning from the tftpboot folder.
I would think it would be very easy to verify FreePBX is writing the correct info to the correct files (if I knew where the files were) and then do a verify/debug as needed and as suggested in the spec.
I would also think that these details should be clearly documented somewhere in the FreePBX organization. Does anyone have URL I can try? Is this something FreePBX does not want known because of all the hacking activity from Asia and the Middle East? All hints welcome.