Hi William,<br><br>My DID is working and am receiving calls. The problem is with Teliax settings from their end. Thank you for spending your valuable time for me. <br><br>Regards,<br>Chandra.<br><br><b><i>William Piper <william.piper@gmail.com></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> <div>Your server seems to be doing exactly what you are telling it to do:</div> <div> </div> <div> -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack<br> -- Playing 'ss-noservice' (language 'en')<br> </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> </div> <div>bp<br><br> </div> <div><span class="gmail_quote">On 10/8/06, <b
class="gmail_sendername">Crazy Boy</b> <<a href="mailto:crazymoonboy@yahoo.com">crazymoonboy@yahoo.com</a>> wrote:</span> <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">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>-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/216.89.79.2-09e1d020", "") in new stack<br> -- Executing Wait("SIP/216.89.79.2-09e1d020", "2") in new stack <br> -- Executing Playback("SIP/216.89.79.2-09e1d020", "ss-noservice") in new stack <br> -- Playing 'ss-noservice' (language 'en')<br> -- Executing Congestion("SIP/216.89.79.2-09e1d020", "") 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", "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", "s|1") in new stack <br> -- Goto (from-sip-external,s,1)<br> -- Executing GotoIf("SIP/216.89.79.2-09e1d020", "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/216.89.79.2-09e1d020", "") in new stack <br> == 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. Please tell me the solution. Looking forward to your response. Thank you. <br><br>Regards,<br>Chandra.</div><span class="ad"> <div> </div><hr size="1"> Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://messenger.yahoo.com" target="_blank"> Great rates starting at 1¢/min. </a>
<div></div><div></div></span><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://easynews.com/" target="_blank">Easynews.com </a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank"> http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br><br></blockquote></div><br> _______________________________________________<br>--Bandwidth and Colocation provided by Easynews.com --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users<br></blockquote><br><p> 
<hr size=1>Stay in the know. Pulse on the new Yahoo.com. <a href="http://us.rd.yahoo.com/evt=42974/*http://www.yahoo.com/preview"> Check it out.</a>