No. It may work if you configure VoIP and intranet to be one network, but I can’t give guarantees without looking deep into the source code.
Merging the VoIP and internet can only be done by your VoIP provider.
What if we look at this from another angle: As it stands the remote device registers fine and receives calls just fine as well as incoming audio works just fine, the issue is there is no outgoing audio from the device and because of the Invite problem the call terminates after 20 seconds.
I have UDP ports 5-65000 open and fowarded to 192.168.1.64, which is the Asterisk server on the internal network.
Any ideas?
Also, just to let you know, my SIP service provider refuses to change my sip IP to my public IP because its against their policy to open their SIP service to the Internet.
One other idea, I read about hosts in a post, here is how my /ect/hosts file is configured:
IPv4 and IPv6 localhost aliases
127.0.0.1 localhost
::1 localhost
192.168.1.64 “name of my server”
Is this correct?
So I solved this…kind of.
I set up a new FreePBX Asterisk Server (CentOS, Distro) and pointed the external IP to that. Then I created a SIP trunk between the “internal” server (server with the SIP service) and the “external” server (which connects to the external devices). The external server makes sends all calls to the internal and the internal sends the call out of the SIP trunks. Hope this helps someone out.