Yes I have searched…
I found similar but not the same as what I need.
Not super familiar with Asterisk, FreePBX, or Linux… and go!
So we are switching from AT&T PRI handoff to a SIP handoff so we can virtualize and create redundancy. Also looking to possibly drop AT&T for out VOIP (been eyeballing SIPstation)…
Just rebuilt a server today, it’s about 15 hours old. Originally setup to connect to the LAN and go through the firewall into the AT&T router with SIP on the incoming port. After the network guy fiddled with the firewall for a good 1.5 hours. NAT, ACL, outboung NAT… all setup and my audio is great going out but coming in is well…crap.
So AT&T tech suggests setting up a second NIC on the PBX to connect to a SIP only connection on their router.
I’ve setup the second NIC for the guest (oh we are using this on a VMWare ESXi 5.5) and from the FreePBX web console I configured it with the IP they gave me to use (GW and mask as well). Checking it from SSH it shows both eth1 and eth2 up and running (eth0 died when I had to change the MAC for a reserved lease from the DHCP server).
Things looking up I think… Dial a call… nothing. Try calling in… nothing. Pulling up the Asterisk CLI to watch the errors and such…
WARNING[C-00000005]: chan_sip.c:23134 handle_response_invite: Received response: “Forbidden” from ‘“Company” sip:`[email protected]`;tag=as6a476494’
*phone number and IP changed to protect the innocent.
So looking here and there… I’m seeing things about how you need to set the trunk to use the correct interface for SIP connections to provider… blah blah blah… but nothing seems to be resolved for any of those users and I’m not finding any definitive answers.
Keep in mind that I still will have internal SIP comms to the phones on the other NIC and it will also have tunnelled traffic to our other office with and IAX2 connection.
Any ideas, suggestions, questions. If you need info please let me know how to obtain it for you.
Thanks to any and all help.