On Mon, Apr 23, 2007 at 06:18:32PM +1000, Daniel Pittman wrote: > G'day. > > I am having reasonable success getting Asterisk 1.4.2 running and doing > what I want, but I can't figure out one particular idiom that I want: > > There are a few situations where I want to have Asterisk push a call > through to the first available transport on a list, such as: > > I have two SIP ports attached to one local (two port) analog phone > system. I want to ring line 1 for the first call, line 2 for the second > call and go to voicemail for the third and subsequent. > > I can't work out the best way to express that. > > Using "Dial(SIP/line1&SIP/line2)" will ring both lines at the same time > which is not really what I want.
You might want to look at doing this with a queue, and then directing the call into the queue. There are some new queue strategies in 1.4.x that might do what you want, and it also has "autofill" option which might make it behave the way you want. There is also a "linear" type strategy which looks like it is making its way into the code, which might be more suitable than roundrobin/rrmemory. http://bugs.digium.com/view.php?id=7279 Or, you might be able to implement it by using the ChanIsAvail command in the dialplan (If the device is returning reasonable things.) It can be used to test availability of a channel or a list of channels and returns the status, or the available channel name. I do a similar thing here and it works very well. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ [EMAIL PROTECTED] - tel: +44 (0)20 7645 3510 - RL786-RIPE _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
