On 12/29/2009 1:01 AM, Jeremy Kister wrote: > e.g., in the first call, below, the channel name is > "SIP/vgw1-00000075" -- the second call (on the same FXO port after a > soft hangup on the CLI) is "SIP/vgw1-00000077" > > How can I extract this information in the dialplan so that I can use > the SoftHangup app in asterisk to disrupt an existing call ?
can anyone think of a different mailing list which might have members who know the answer i'm looking for? asterisk-dev ? -- Jeremy Kister http://jeremy.kister.net./ _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
