<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Mark,<div><span class="Apple-tab-span" style="white-space:pre"> </span>You should join both #openzap and #freeswitch on irc.freenode.net there are way too many things to go over and the list would just be too slow.</div><div><br></div><div>/b</div><div><br><div><div>On Mar 17, 2009, at 5:35 PM, Mark Thomas wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; ">Hello, everyone.<br><br>I am new to Freeswitch, and telephony in general. I am trying to set up a Freeswitch system at work for a project, and I have hit a wall.<br><br>I have a dedicated LD T1 from Qwest and a Sangoma A104 card. I believe I have openzap correctly installed in wanpipe mode. I am trying to bridge an incoming SIP call from an IP phone to an openzap channel without success. The Freeswitch log shows that dialing takes place, but the call never completes.<br><br>The call log is here:<span class="Apple-converted-space"> </span><a href="http://pastebin.freeswitch.org/7805">http://pastebin.freeswitch.org/7805</a><br>The dialplan xml, openzap.conf, and openzap.conf.xml are here:<span class="Apple-converted-space"> </span><a href="http://pastebin.freeswitch.org/7806">http://pastebin.freeswitch.org/7806</a><br><br>Any help greatly appreciated.<br><br>--Mark<br><br>______________</span></blockquote></div><br></div></body></html>