Based on my experience I would say that using ${DIALSTATUS} variable would be the most common way to do it...
On 4/23/07, Daniel Pittman <[EMAIL PROTECTED]> 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. Using two sequential Dial() commands into the extension will ring the lines one after the other -- even if it times out on the first line, which is again not what I want. At the moment my best guess is that I need to use the DIALSTATUS variable and implement the fail-over process based on that. That seems cumbersome, though -- surely this isn't a terribly uncommon requirement? Regards, Daniel -- Digital Infrastructure Solutions -- making IT simple, stable and secure Phone: 0401 155 707 email: [EMAIL PROTECTED] http://digital-infrastructure.com.au/ _______________________________________________ --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
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