Asterisk21 and FreePBX17 were installed in VMware Workstation(Ubuntu24.04). The MicroSIP software was used for testing, and successful communication was achieved, but there was no sound.
This is the status of my port number availability.
Asterisk 21 doesn’t include chan_sip, so sip.conf will be completely ignored. Also, you should only ever update .conf files that have “custom” in their name, on FreePBX.
externip and localnet have GUI options in FreePBX 17.
nat=yes has individual settings, but both are intended for the case where the peer is behind NAT, and Asterisk is not behind (the same) NAT.
RTP Is UDP, so there is no need to open 10000 to 20000 TCP.
Where is MicroSIP in relation to your routers, and are there any NAT routers between any of its instances and the FreePBX machine?
Can you confirm that you were only attempting local calls, as you don’t mention any provider trunks.
We’ll almost certainly need the “pjsip set logger on” type logs, to see what is in the SDP.
MicroSip is located on the machines at 192.168.3.39 and 192.168.3.71. There is a router between these two machines. The FreePBX server is on the virtual machine of the machine at 192.168.3.39.
Yes, I was just trying to make a local call.
There wouldn’t normally be any NAT involvement in such a case. Can you confirm that the router can route between all the addresses involved?
Although it is wrong to have non-zero bits outside the network mask, I’m fairly sure that Asterisk copes with them. However, it might be best to fix that and replace the last 1 in your local networks with a zero.
Yes, the router can route to all relevant addresses. The subnet mask has been modified, but there is still no sound in the calls, and video calls also fail to display the video.
In that case, I think we need RTP debugging enabled, and also the full log, with the CLI command “pjsip set logger on” in effect (this resets on a restart).
