Hi all,
Does Asterisk 11.6 + WebRTC module support hold/unhold of WebRTC call? I mean I use browser (Chrome 31) and latest JsSip (from hold/unhold branch) to instantiate the call but for some reason I can’t hold/unhold calls.
Developers of JsSip state:
- I have some doubts because sipml5.org allowed me to hold/unhold my call with all the same settings I use with JsSip.