<div>Hello Chandra,</div>
<div> </div>
<div>What about Teliax´s service? Is it recommended? How´s their call quality? Thanks in advance...</div>
<div> </div>
<div><br><br> </div>
<div><span class="gmail_quote">On 10/10/06, <b class="gmail_sendername">Crazy Boy</b> <<a href="mailto:crazymoonboy@yahoo.com">crazymoonboy@yahoo.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">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.
<span class="q"><br><br>Regards,<br>Chandra.<br><br><b><i>William Piper <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:william.piper@gmail.com" target="_blank">william.piper@gmail.com</a>></i>
</b> wrote:</span>
<div><span class="e" id="q_10e307fabe2e238f_2">
<blockquote style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(16,16,255) 2px solid">
<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 onclick="return top.js.OpenExtLink(window,event,this)" href="http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf" target="_blank">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 onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:crazymoonboy@yahoo.com" target="_blank">crazymoonboy@yahoo.com
</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: rgb(204,204,204) 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">"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>
<div></div>
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