Hello,
First I need to say that I am not a SIP expert. Forgive me if my question is not clear enough.
I am trying to setup MicroSIP to act as a phone extension on our company FreePBX / Asterisk server. I am running microsip on my home computer which is connected to the office network via openvpn (pfSense).
I have created a new extension in freepbx with user / password and entered that in microsip . Microsip says that it is OnLine. From that point, I can call my cell phone using microsip as I would from within the office and I can also call that extension from my cell phone. However, microsip does not receive any audio but it can transmit ( microphone working fine). I do however hear the microsip phone ringing when I get a call.
I suspected a firewall issue or UPD packets being blocked. I turned off windows firewall and defender.
I also setup the asterisk extension to use only TCP only for the transport. From the asterisk server, I can connect to port 5060 on my home PC. I also tried to change the codecs supported by microsip in case that would be an issue.
I tried to look in asterisk log but I cannot see any obvious issues with connection during the calls but I am not too sure what to look for. I know that there are also the RTP ports (10000 - 20000 ) to check for but I dont know how to check them. I do not see any listening ports in the range on my home PC.
My home pc is connected to the office via a vpn so my ip is on a different subnet but the route seem to work both ways ( I can ping my pc from asterisk and vice versa).
Our installation is pretty old: we have FreePBX 2.10 and asterisk 1.8.15.1 .
What could be the issue and how can I run more informative tests ? Any hint would be useful
Thank you
Gilbert