Closed deleolajide closed 5 years ago
In version 2.0 and going forward, all SIP and legacy telephony is implemented by an outgoing call to a PBX (Asterisk or FreeSWITCH) using the audioconf plugin for Converse. A conference should be either in an SFU (Jitsi) or MCU (telephony). The complexity of bridging both has been dropped.
If a bridge is required between an SFU conference and a telephone MCU bridge, please use the public meet.jit.si web site.
Pàdé should auto-discover the jigasi plugin for openfire and configure jitsi-meet to support telephone-based participants. It should: