I’have recently deployed FreePBX 15 with Asterisk 16 and I have odd issues. When click on “Apply Config” sometimes it takes too long and then shows xhr empty respoonse error message which never seen in FreePBX 13. Also, it seems it does not configure asterisk properly because same config I used on FreePBX 13 does not work on 15. For example, I have deleted a trunk and when run asterisk -x “sip show peers”, it still shows deleted trunk.
Also I have some issues such sip accounts always raise forbidden 403 error even when username and secret is correct. When I set “Allow Annonymus” to enabled, incoming calls from trunk works but if I turn it off, it does work. I have used exactly same config for the trunk as I was using on 13. Bellow is my sip settings for the trunk:
Looks to me like you’ve created a duplicate peer by putting the same information in both Outgoing and Incoming sections of the chan_sip trunk config. Just use the Outgoing section.
I don’t know if this even works in Asterisk 16, but it’s been deprecated a very long time.
I have remove insecure and authencticate and changed transport to udp in outgoing section and cleared incoming section. Now outgoing calls are established but no audio is streamed. Both SIP phone and FPBX are behind NAT and external IP and local network is detected correctly. I have already forwarded following ports in my TD-LTE modem/router.
5060-5061 TCP/UDP
10001-20000 UDP (Port 10000 cannot be forwarded on this specific modem, RTP range has alreadt be configured to use this range)