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;">"The number you have dialed is not inservice"</span>. Here I am giving the output from Asterisk server console: <br><br>*CLI><br> -- IAX2/teliax-2 answered SIP/350-09e3b540<br> -- Executing GotoIf("SIP/216.89.79.2<div id="mb_0"><wbr>-09e1d020", "0?from-trunk||1") in new stack<br> -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack <br> -- Channel will hangup at 2006-10-06 11:27:55 UTC.<br> -- Executing Answer("SIP/216.89.79.2<wbr>-09e1d020", "") in new stack<br> -- Executing Wait("SIP/216.89.79.2-09e1d020<wbr>", "2") in new stack <br> -- Executing
Playback("SIP/216.89.79.2<wbr>-09e1d020", "ss-noservice") in new stack<br> -- Playing 'ss-noservice' (language 'en')<br> -- Executing Congestion("SIP/216.89.79.2<wbr>-09e1d020", "") in new stack <br> == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/216.89.79.2-09e1d020'<br> -- Executing NoOp("SIP/216.89.79.2-09e1d020<wbr>", "Hangup") in new stack<br> -- Executing Set("SIP/216.89.79.2-09e1d020", "DID=s") in new stack <br> -- Executing Goto("SIP/216.89.79.2-09e1d020<wbr>", "s|1") in new stack<br> -- Goto (from-sip-external,s,1)<br> -- Executing GotoIf("SIP/216.89.79.2<wbr>-09e1d020", "0?from-trunk|s|1") in new stack <br> -- Executing Set("SIP/216.89.79.2-09e1d020", "TIMEOUT(absolute)=15") in new stack<br> -- Channel will hangup at 2006-10-06 11:28:04 UTC.<br> -- Executing
Answer("SIP/216.89.79.2<wbr>-09e1d020", "") in new stack <br> == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/216.89.79.2-09e1d020'<script><!-- D(["mb","<br><br>I tried in changing the "Features" on my account page. But, no use. Here I am enclosing my configuration files also. Please solve my problem. Looking forward to your response. Thank you.\n<br><br>Regards,<br>Chandra.<br>\n\n",0] ); //--></script><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. Please tell me the solution. Looking forward to your response. Thank you. <br><br>Regards,<br>Chandra.</div><p> 
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