<div>Hi</div>
<div> </div>
<div>My suggestion is </div>
<div>run Ngrep is the best tool to see what is happening</div>
<div>is the DID calling from PSTN, the call is landing in the server or not</div>
<div> </div>
<div>So better option is that to check with Ngrep DID will give you details</div>
<div> </div>
<div>Ram<br><br> </div>
<div><span class="gmail_quote">On 8/15/06, <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">
<div>Hi,<br><br>Thank you very much for your patience to give solutions for me. Today is holiday for us because of our Independance day. Tomorrow I will do and check as suggested by you and let you know. <br><br>Once again, Thank you.
<br><br>Regards,<br>Chandra.</div>
<div><span class="e" id="q_10d10a0b0f07f9f3_1"><br><br><b><i>Lacy Moore - Aspendora <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:aspendora@gmail.com" target="_blank">aspendora@gmail.com</a>>
</i></b> wrote:</span></div>
<div>
<blockquote style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: rgb(16,16,255) 2px solid"></blockquote></div>
<div><span class="e" id="q_10d10a0b0f07f9f3_3">
<div>I struggled with one provider for a long time until finally realizing my username on their site was not my username that I was supposed to be using in sip configuration. Make sure you are using the right username and password. However, it would seem that you would not be able to make an outgoing call using the wrong username/password combination.
</div>
<div> </div>
<div>One thing I have not seen in your posts is your firewall information. Your firewall may be setup to allow outgoing connections, but not incoming. I would not depend on info from a provider. You may very well be registering with them, but your firewall may be blocking the incoming call. If you think you have no firewall, check again. IPTABLES might have loaded itself and it may be blocking. Try:
</div>
<div> </div>
<div>service iptables stop</div>
<div> </div>
<div>and then try the incoming call again. I've been burned twice due to this. Something has changed in the way I configure my linux boxes, and for some reason iptables is starting.<br><br> </div>
<div><span class="gmail_quote">On 8/14/06, <b class="gmail_sendername">Rich Adamson</b> <<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:radamson@routers.com" target="_blank">radamson@routers.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">Crazy Boy wrote:<br>> Hi,<br>><br>> Thank you for your response. As you said, I executed the command "sip
<br>> show registry". But, its not showing anything. Teliax people are also<br>> telling that my Asterisk server doesn't register with Teliax. So, the<br>> final conclusion is "My Asterisk server doesn't register with Teliax".
<br>> Here I am giving my configuration files. Now, What I have to do to<br>> register my Asterisk server with Teliax? Please tell me.<br>><br>> SIP.CONF contents:<br>><br>> [general]<br>> register =>
<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:xyz.abc:xxxxxxx@voip-co1.teliax.com" target="_blank">xyz.abc:xxxxxxx@voip-co1.teliax.com</a><br>> [authentication]<br>> auth = <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:xyz.abc:xxxxxxx@voip-co1.teliax.com" target="_blank">
xyz.abc:xxxxxxx@voip-co1.teliax.com </a><br><br>Double check the above two statements to ensure the userid and password<br>are exactly those provided to you by teliax. There is nothing else in<br>your config that impacts the register statement with the exception of
<br>nat'ing.<br><br>It would appear from your other config statements that asterisk might be<br>located behind a firewall or nat box. If so, read the documentation on<br>that, and look at the asterisk/configs/sip.conf.sample file.
<br>Specifically the section on "NAT SUPPORT".<br><br>You might also want to read more about using the diagnostic tools<br>available to you within asterisk. Setting verbose and/or debug to a high<br>number and copy/paste the CLI output associated with the problem. Or,
<br>start using the CLI with something like:<br> asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv<br><br>> [teliax-incoming]<br>> exten => 3031234567, 1, Answer()<br>> exten => 3031234567, 2, Dial(SIP/105,15)
<br><br>The above has nothing to do with registering with teliax, but you do not<br>want to "answer" a call before ringing the sip phone. Take that<br>statement out of there. When the sip phone answers an incoming call,
<br>asterisk will automatically send the answer to teliax.<br><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></blockquote></div><br><br clear="all"><br>-- <br>Lacy Moore<br></span></div>
<div>Aspendora, Inc. _______________________________________________</div>
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