<div class="gmail_quote">On Wed, Dec 8, 2010 at 1:50 PM, Tilghman Lesher <span dir="ltr"><<a href="mailto:tlesher@digium.com">tlesher@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div class="im">On Wednesday 08 December 2010 11:33:10 Bruce B wrote:<br>
> Thanks for the input guys. I really appreciate all the input and I am<br>
> sure they work but I thought there would be a much better way to do<br>
> this. Sounds like patching things to me. Why doesn't Asterisk take<br>
> advantage of the qualify values to make sure if the SIP connection is<br>
> up or not? Shouldn't this become a native feature of the PBX rather<br>
> than trying to do work-around?<br clear="all"></div></blockquote></div><br>Enable qualify on your SIP peers. Then configure your outbound dialplan like this:<br><br>exten => outboundexten,1,Dial(SIP/...)<br>exten => outboundexten,n,Dial(DAHDI/...)<br>
<br>If the SIP is up, the call goes through on that Dial statement. When the call is over and the people hangup, it will not continue down to the next line. If it's down (and the system knows this because of qualify), it will fail and automatically roll to the next line. I've used this to setup failover between DAHDI and SIP a few times.<br>
<br>-- <br>Thanks,<br>--Warren Selby, dCAP<br><a href="http://www.selbytech.com" target="_blank">http://www.selbytech.com</a><br>
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