> You can do this in a couple of ways, using the Manager interface > and the "Connect" command. Alternatively, you can create a call > file in Asterisk's call spool (usually /var/spool/asterisk or > whatever) which has the same makeup as the Connect command. We won't be originating any calls out (apart from the queued calls).
The scenario is that we have a number of incoming calls, and some of those are selected to then queue, to be dealt with by people at a remote location. The number of calls queueing is greater than the number of PSTN lines available at the remote site. I *think* it'll work as I described, having a "member => SIP/[EMAIL PROTECTED]" entry in queues.conf? When all lines are busy, the next-in-queue caller will keep retrying and getting a busy result back? Then once a line becomes available the call will go through? Paul _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
