<html>
<head>
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
</head>
<body bgcolor="#FFFFFF" text="#000000">
<br>
<br>
michael k wrote:
<blockquote
cite="mid:CAJm-Ax_xzbCcJY-3BVA2wzLxDAEUOMku2LrefgWwS9GCsPnzUQ@mail.gmail.com"
type="cite">Thanks for the update. but how do i resolve this issue
? can you help me please ? <br>
<br>
<br>
</blockquote>
Can you PLEASE take this to the FreePBX support group?<br>
<br>
It seems obvious to most that therein lies the problem<br>
You are thinking you wish to dial out through the X100, but Asterisk
is attempting to dial out on a non existent SIP connection<br>
Something isn't right in your dialplan, created by FreePBX<br>
<br>
<br>
Also, no echo canceller on the X100 card isn't wise, but you will
not realize that until you are able to use it!<br>
<br>
John Novack<br>
<br>
<blockquote
cite="mid:CAJm-Ax_xzbCcJY-3BVA2wzLxDAEUOMku2LrefgWwS9GCsPnzUQ@mail.gmail.com"
type="cite"><br>
<div class="gmail_quote">On Thu, Sep 29, 2011 at 1:00 PM, Sam
Govind <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex;">Actually
its easier. I haven't worked on FreePBX lately so what I
remember is here: You've created a Zap trunk. Focus on the
Dial Rule - You can keep it empty as well. Then you've created
an outbound route its dial-rule is important.
<div>
<br>
</div>
<div>But the funny thing which I didn't mention before is that
you've ZAP defined in FreePBX but actually its DAHDI so I
remember they've this cute parameter in amportal.conf which
tells FreePBX to convert ZAP into DAHDI. </div>
<br>
</blockquote>
</div>
</blockquote>
<snip><br>
<pre class="moz-signature" cols="10000">--
Dog is my Co-pilot</pre>
</body>
</html>