<span style="font-weight: bold;">Based on my experience I would say that using ${DIALSTATUS} variable would be the most common way to do it...</span><br><br><br><div><span class="gmail_quote">On 4/23/07, <b class="gmail_sendername">
Daniel Pittman</b> <<a href="mailto:daniel@rimspace.net">daniel@rimspace.net</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
G'day.<br><br>I am having reasonable success getting Asterisk 1.4.2 running and doing<br>what I want, but I can't figure out one particular idiom that I want:<br><br>There are a few situations where I want to have Asterisk push a call
<br>through to the first available transport on a list, such as:<br><br>I have two SIP ports attached to one local (two port) analog phone<br>system. I want to ring line 1 for the first call, line 2 for the second<br>call and go to voicemail for the third and subsequent.
<br><br>I can't work out the best way to express that.<br><br>Using "Dial(SIP/line1&SIP/line2)" will ring both lines at the same time<br>which is not really what I want.<br><br>Using two sequential Dial() commands into the extension will ring the
<br>lines one after the other -- even if it times out on the first line,<br>which is again not what I want.<br><br><br>At the moment my best guess is that I need to use the DIALSTATUS<br>variable and implement the fail-over process based on that. That seems
<br>cumbersome, though -- surely this isn't a terribly uncommon requirement?<br><br>Regards,<br> Daniel<br><br>--<br>Digital Infrastructure Solutions -- making IT simple, stable and secure<br>Phone: 0401 155 707 email:
<a href="mailto:contact@digital-infrastructure.com.au">contact@digital-infrastructure.com.au</a><br> <a href="http://digital-infrastructure.com.au/">http://digital-infrastructure.com.au/</a><br><br>_______________________________________________
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