Hi Jay,<br><br>Skype can be used successfully with the ChanSkype module on supported platforms (Fedora Core 3, 4 or 5 or Ubuntu 6.04). It's $19USD for a single personal license, and tends to work quite well. It's not the easiest item to setup (the OS needs a window manager running on it, and each Skype channel require it's own user with it's own desktop session running for that user), but once you get it going I've rarely found it to fail. I used it with the unlimited outgoing calling to North America from Skype, and it's saved me quite a bit.
<br><br>AR<br><br><div><span class="gmail_quote">On 8/15/07, <b class="gmail_sendername">Jay R. Ashworth</b> <<a href="mailto:jra@baylink.com">jra@baylink.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;">
I've looked around a bit, and I'm still not sure I quite know what the<br>state of the union is with regard to configuring SkypeIn/Out and Vonage<br>services as trunk-side appearances on an Asterisk PBX?<br><br>Any good clear pointers?
<br><br>Cheers,<br>-- jra<br>--<br>Jay R. Ashworth Baylink <a href="mailto:jra@baylink.com">jra@baylink.com</a><br>Designer The Things I Think RFC 2100
<br>Ashworth & Associates <a href="http://baylink.pitas.com">http://baylink.pitas.com</a> '87 e24<br>St Petersburg FL USA <a href="http://photo.imageinc.us">http://photo.imageinc.us</a>
+1 727 647 1274<br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.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>Alex Robar<br><a href="mailto:alex.robar@gmail.com">alex.robar@gmail.com</a>