We have a Tandberg VCS System for Video conferencing and a customer running
AsteriskNow (Asterisk 1.6 + FreePBX) for Audio conferencing.
How do we integrate the 2 systems such that Audio SIP calls are seamlessly
passed between the two. Sorry we’re just starting up so a bit of general
advice, or a link to any document would be great!
If anybody has done this – would appreciate any tips 🙂