I’ve just set up a FreePBX server using the distro with Asterisk 11. I’m trying to set up the simplest possible system. I added two extensions - 101 and 102. I’m connecting 101 to the Media5-Fone app on my iPhone. I’m connecting 102 to a ExpressTalk PC soft phone on a Windows PC. When I created the extensions, I changed as little as possible from the defaults - just set the extension, display name, and secret.
Both extensions are connecting to the server via WiFi. Both are on the same access point.
Both can register on the server.
102 can ring 101, but 102 can not ring 101.
102 can transmit audio to 101, but 101 can not transmit audio to 102.
Wah? Any suggestions as to what I should be looking at? I tried a the Zoiper app on my iPhone as well, and it showed the same behavior.
It might well be carrier dependent, many carriers suppress SIP traffic.
The server and both extensions are on my intranet and all on the same subnet. Nothing is touching the internet.
The audio is two RTP streams they must agree on the port used, and nothing in your router/gateway/wifi box should change those ports.
You can see the rtp streams in asterisk with
rtp set debug ip|on|off
you can also use tcpdump on the network to see what is being mangled.
And why one extension cannot call another with, from asterisk
sip set debug ip|on|off|peer
will reveal the SIP “conversation” or not.
Thanks for the help. Turned out that due to a Windows Update gone wrong, something got munged in my registry that resulted in Windows Firewall overstepping its bounds. All is now well - rings and audio in both directions.