Hi All, We have a Tandberg VCS System for Video conferencing and a customer running AsteriskNow (Asterisk 1.6 + FreePBX) for Audio conferencing.
Problem Statement: 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! In asterisknow, we basically created an outbound route as well as a SIP trunk pointing to the Tandberg VCS as follows: Outgoing Settings: host=192.168.1.X username=username secret=******** type=peer Incoming Settings: User Context: vcs secret=****** type=user context=from-trunk Not sure if above is correct, and what to add at the Tandberg side to make this work as there are limited doc from Tandberg. If anybody has done this - would appreciate any tips :) Thanks! Jake
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