yes.. actualy use 1 did for each proxy to check..<br><br>then inbound for each use the method he described..<br><br><br><div><span class="gmail_quote">On 10/12/06, <b class="gmail_sendername">Mojo with Horan & Company, LLC
</b> <<a href="mailto:mojo@horanappraisals.com">mojo@horanappraisals.com</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;">
on an analog Zap PSTN channel, you have no real way of determining if<br>the remote side answered, because, as you discerned, it IS considered<br>answered as soon as asterisk opens the channel.<br><br>How about you contact another asterisk server through the PSTN, and dial
<br>through to an extension on that remote asterisk server that, in turn,<br>notifies the first asterisk server maybe via the internet that it was<br>received?<br><br>for example, consider the following php script accessupdate.php
on<br>primary asterisk box:<br><br><?php<br> if (!strcmp($_GET['update'], 'true'))<br> {<br> touch("/etc/asterisk/secondary_server_last_access");<br> }<br>?><br><br>then primary calls secondary box through PSTN, and through the magic of
<br>DISA or CID or what-have-you, dials through to an extension that executes<br>System(wget -q -O /dev/null<br><a href="http://primary-server/access_update.php?update=true">http://primary-server/access_update.php?update=true
</a>)<br><br>then hangs up. then primary server checks the last-access time of<br>/etc/asterisk/secondary_server_last_access to make its decision, via<br>cron script or bash script triggered through the dialplan subsequent to
<br>the initial dial-out.<br><br>This is of course a very rudimentary on-the-fly thing I came up with,<br>but think outside the box and this may be the easiest way for you to do<br>what you want.<br><br>Moj<br><br><br>John Kane wrote:
<br>> I am trying to write a script to attempt to make a call on a Zap<br>> channel, and if it fails, send an alarm. I can generate the call, but<br>> because the Zap channel accepts the call, even though the other end
<br>> never answers, it sees it as a successful call, which it isn't.<br>><br>><br>><br>> Anyone have any ideas on this? Thanks.<br>><br>> !DSPAM:500,452d7fa8199221504517840!<br>><br>><br>> ------------------------------------------------------------------------
<br>><br>> _______________________________________________<br>> --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>><br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:
<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>><br>><br>> !DSPAM:500,452d7fa8199221504517840!<br><br>--<br>Mojo <<a href="mailto:mojo@horanappraisals.com">
mojo@horanappraisals.com</a>><br>Office Manager, Horan & Company, LLC<br>(907) 747-6666 x112<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">
Easynews.com</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users
</a><br></blockquote></div><br><br clear="all"><br>-- <br>Mike<br>Sales Manager<br><a href="http://www.theclubvoip.com">http://www.theclubvoip.com</a><br>Making it happen<br>1.877.807.VOIP (8647)