I have 2 servers 1st has ip 10.0.1.60 2nd 10.0.31.60
Im trying to setup a communication with 2 virtual FreePBX and tight them up using trunks.
Communication between servers seems to be established successfully after setting up trunks but every time im trying to call from 1st FreePBX to 2nd I get a response in the phone “all lines is busy right now please try your call again later”.
“trunk” is not an Asterisk or SIP term, it is a FreePBX one.
You are using an obsolete channel driver, which is not in the current development branch of Asterisk.
type=friend is a security and misoperation risk. Also, in cases where its use is valid, it is impossible to check without the section name. Use type=peer.
insecure=invite has no effect, as there is no secret. If there were a secret, it would make things insecure without giving any benefit.
insecure=port is insecure and gives no benefit in an Asteirsk to Asterisk case with no NAT.
The default (auto) setting for nat is perfectly adequate, although explicitly disabling force_rport and comedia does no harm here.
trunk is not an option that is recognized by chan_sip.
You don’t say from which system the logs were taken, although as it only shows an IAX2 channel, it has to be the first one.
This is the wrong forum to have FreePBX logs analyzed, but I would say that it thought the destination was invalid before it actually tried to Dial it; it is probably trying to access the device state, but we can’t tell that just from the logs. Could be wrong name. Could be lack of connectivity. To get your FreePBX log analyzed, you could try https://community.freepbx.org/