Thanks for the input guys. I really appreciate all the input and I am sure they work but I thought there would be a much better way to do this. Sounds like patching things to me. Why doesn't Asterisk take advantage of the qualify values to make sure if the SIP connection is up or not? Shouldn't this become a native feature of the PBX rather than trying to do work-around?<div>
<br></div><div>Thanks<br><br><div class="gmail_quote">On Wed, Dec 8, 2010 at 12:20 PM, <span dir="ltr"><<a href="mailto:klitzing@pool.informatik.rwth-aachen.de">klitzing@pool.informatik.rwth-aachen.de</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi!<br>
<div class="im"><br>
> There are situations when internet connection is lost, SIP provider<br>
> fails, or even authentication to SIP provider fails, and we want to use<br>
> the backup Dahdi channels (PSTN). As simple as it may sound but with<br>
</div>> the manydifferentsituations and error messages it seems like it's not<br>
<div class="im">> so easy to predict all the errors. Is there any single parameter value<br>
> that can be changed to send the call to Dahdi instead of SIP<br>
<br>
</div>There is nothing available out-of-the-box. You need to include your own IP & SIP tests in the<br>
dialplan before dialing out to a SIP channel. Useful for this purpose are<br>
<br>
- ping and host or wget,<br>
- GROUP() and GROUP_COUNT(),<br>
- SIPPEER(xxx:status),<br>
- CHANISAVAIL(),<br>
- dial timeouts and<br>
- post-dial error handling (see DIALSTATUS and HANGUPCAUSE as well as Asterisk 1.8<br>
with its ability to act directly upon the SIP response code).<br>
<br>
Philipp<br>
<div><div></div><div class="h5"><br>
<br>
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