[BUG] dSIPRouter WebRTC to SIP Proxy (FreePBX PJSIP) extension unavail
Opened this issue · 1 comments
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
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.