<div>Your server seems to be doing exactly what you are telling it to do:</div>
<div>&nbsp;</div>
<div>&nbsp;-- Executing Playback(&quot;SIP/216.89.79.2-09e1d020&quot;, &quot;ss-noservice&quot;) in new stack<br>&nbsp;-- Playing 'ss-noservice' (language 'en')<br>&nbsp;</div>
<div>Read the extensions.conf directions on the wiki site:</div>
<div><a href="http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf">http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf</a></div>
<div>&nbsp;</div>
<div>bp<br><br>&nbsp;</div>
<div><span class="gmail_quote">On 10/8/06, <b class="gmail_sendername">Crazy Boy</b> &lt;<a href="mailto:crazymoonboy@yahoo.com">crazymoonboy@yahoo.com</a>&gt; wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Hi,<br><br>I have created SIP extenstions and created Teliax Trunk using IAX2. I am making outgoing calls to USA successfully.
<br><br>When I am making a call to my DID number from outside, its telling that <span style="FONT-WEIGHT: bold">&quot;The number you have dialed is not inservice&quot;</span>. Here I am giving the output from Asterisk server console: 
<br><br>*CLI&gt;<br>&nbsp;&nbsp;&nbsp; -- IAX2/teliax-2 answered SIP/350-09e3b540<br>&nbsp;&nbsp;&nbsp; -- Executing GotoIf(&quot;SIP/216.89.79.2 
<div>-09e1d020&quot;, &quot;0?from-trunk||1&quot;) in new stack<br>&nbsp;&nbsp;&nbsp; -- Executing Set(&quot;SIP/216.89.79.2-09e1d020&quot;, &quot;TIMEOUT(absolute)=15&quot;) in new stack <br>&nbsp;&nbsp;&nbsp; -- Channel will hangup at 2006-10-06 11:27:55 UTC.
<br>&nbsp;&nbsp;&nbsp; -- Executing Answer(&quot;SIP/216.89.79.2-09e1d020&quot;, &quot;&quot;) in new stack<br>&nbsp;&nbsp;&nbsp; -- Executing Wait(&quot;SIP/216.89.79.2-09e1d020&quot;, &quot;2&quot;) in new stack <br>&nbsp;&nbsp;&nbsp; -- Executing Playback(&quot;SIP/216.89.79.2-09e1d020&quot;, &quot;ss-noservice&quot;) in new stack
<br>&nbsp;&nbsp;&nbsp; -- Playing 'ss-noservice' (language 'en')<br>&nbsp;&nbsp;&nbsp; -- Executing Congestion(&quot;SIP/216.89.79.2-09e1d020&quot;, &quot;&quot;) in new stack <br>&nbsp; == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020'
<br>&nbsp;&nbsp;&nbsp; -- Executing NoOp(&quot;SIP/216.89.79.2-09e1d020&quot;, &quot;Hangup&quot;) in new stack<br>&nbsp;&nbsp;&nbsp; -- Executing Set(&quot;SIP/216.89.79.2-09e1d020&quot;, &quot;DID=s&quot;) in new stack <br>&nbsp;&nbsp;&nbsp; -- Executing Goto(&quot;SIP/216.89.79.2-09e1d020&quot;, &quot;s|1&quot;) in new stack
<br>&nbsp;&nbsp;&nbsp; -- Goto (from-sip-external,s,1)<br>&nbsp;&nbsp;&nbsp; -- Executing GotoIf(&quot;SIP/216.89.79.2-09e1d020&quot;, &quot;0?from-trunk|s|1&quot;) in new stack <br>&nbsp;&nbsp;&nbsp; -- Executing Set(&quot;SIP/216.89.79.2-09e1d020&quot;, &quot;TIMEOUT(absolute)=15&quot;) in new stack
<br>&nbsp;&nbsp;&nbsp; -- Channel will hangup at 2006-10-06 11:28:04 UTC.<br>&nbsp;&nbsp;&nbsp; -- Executing Answer(&quot;SIP/216.89.79.2-09e1d020&quot;, &quot;&quot;) in new stack <br>&nbsp; == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020'
<br><br>When I am calling from outside phone, call is coming to my server and is not routing. I am making calls to USA and between SIP extensions successfully.&nbsp; Please tell me the solution. Looking forward to your response. Thank you. 
<br><br>Regards,<br>Chandra.</div><span class="ad">
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