Hi Libera,<br><br>We have an account with Teliax from 7 months. Teliax's service is very good and giving excellent customer support also. But, I observed the below things from Teliax's people.<br><br>1) Let us assume that you have configured your Teliax account settings with XLite or any other sofphone directly without using Trixbox or Asterisk. After that, if you are facing any problem, they are solving.<br><br>2) If you configure Teliax account settings with Asterisk or Trixbox, they are facing trouble to solve some technical problems from Trixbox or Asterisk point of view<br><br>3) Voice quality is very good.<br><br>Thank you.<br><br>Regards,<br>Chandra.<br><br><br><br><b><i>"R.R Libera" <astecomm@gmail.com></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> <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="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">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="border-left: 2px solid rgb(16, 16, 255); padding-left: 5px; margin-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 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="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> <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
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