[BUG] dSIPRouter WebRTC to SIP Proxy (FreePBX PJSIP) extension unavail

[gudge25]

Describe the bug dSIPRouter WebRTC to SIP Proxy when I connect from sipML5 extension is connected but asterisk cannot qualify extension so set it as Unavail

To Reproduce Steps to reproduce the behavior: using video https://www.youtube.com/watch?v=nOHwrmuLLL0&t=233s use dSiprouter with Freepbx PJSIP for some reason PJSIP Xten is Unavail

Expected behavior Qualify should work

Server Info:

• OS: debian 11
• Distro: output from cat /etc/os-release
• dSIPRouter Version: output from Version: 0.721

Client Info:

• Device: (https://www.doubango.org/sipml5/call.htm?svn=252#
• OS: MacOS
• Client Software: Chrome 114.0.5735.106

[dmitrii507]

Hi, I have a similar problem, but I had to manually specify the local address for listening in order to configure tls to work. After that the connection started to go through to FreePBX but the status in it is indicated as not on the network. Do you have a server behind NAT? I'm just thinking to place the server not behind nat to check the work.