HI, im trying to make this work, theres the good old problem of one way audio or no audio at all, it is surely a NAT problem acording to what i've seen on the forum, my setup is the following::
- VM with Issabel is on a local network with SIP phones.
- All traffic goes through a CISCO 1921 router.
- My Sip provider does not have a pulic IP, it is conected via fiber to the router with a separate link from the internet, like a "leased line".
- The router forwards ports 5060 and RTP 10.000 to 60.000 to the sip server of our provider (they asked for that range)
- The router NATs traffic from the issabel PBX to the Internet and the CPA (sip provider).
The network is this one: https://drive.google.com/file/d/112-9KUK_gv3hIxwEKPXzsH8wKxHgURNm/view?usp=drivesdk
*When i make calls via the provider, i can hear them but they cant hear me, since i forward port 5060 from the routers public IP to the PBX i can use SIP phones from outside the network, they have two way audio but the call terminates after 15 sec, when i make internal calls in LAN theres no issue.
I have a few public IPs available but i would need to change the network a little in order to user them, if nothing helps, would using a public IP for Issabel fix everything?
