<div>Hi,</div>
<div> </div>
<div>Thanks for your reply<br></div>
<div> </div>
<div>I am using my own number and not hanging up. and sip debug is also not showing much<br>information regarding the failure.<br></div>
<div>please suggest our what might be the problem.</div>
<div> </div>
<div>Any help is highly appreciated.</div>
<div><br>Thanks.</div>
<div><br> </div>
<div class="gmail_quote">On Fri, Apr 24, 2009 at 4:58 AM, Steve Totaro <span dir="ltr"><<a href="mailto:stotaro@totarotechnologies.com">stotaro@totarotechnologies.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid"><br><br>
<div class="gmail_quote">
<div class="im">On Thu, Apr 23, 2009 at 6:12 PM, Matt Riddell <span dir="ltr"><<a href="mailto:lists@venturevoip.com" target="_blank">lists@venturevoip.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<div>On 24/04/2009 3:00 a.m., Sam Hawkin wrote:<br>> Hi All,<br>><br>> I am trying to use the AMD (Answering Machine Detect).<br>> But it is not sending the AMD_Status as either<br>> the Human or Machine, it hangs up in middle.<br>
<br></div>I'd say that the remote end of the call is hanging up - do a SIP debug<br>so you can see what happens - the best way to test things like this is<br>by calling your own number - that way you can guarantee it doesn't hang<br>
up :)<br><br>--<br>Kind Regards,<br><br>Matt Riddell<br>Director<br>_______________________________________________<br><br><a href="http://www.venturevoip.com/" target="_blank">http://www.venturevoip.com</a> (Great new VoIP end to end solution)<br>
<a href="http://www.venturevoip.com/news.php" target="_blank">http://www.venturevoip.com/news.php</a> (Daily Asterisk News - html)<br><a href="http://www.venturevoip.com/newrssfeed.php" target="_blank">http://www.venturevoip.com/newrssfeed.php</a> (Daily Asterisk News - rss)<br>
<div>
<div></div>
<div></div></div></blockquote></div>
<div><br> <br>You can also run Orecx on the localhost (for very small production or lab systems) or on a different host via mirrored switch port and then listen to all calls (SIP and other VoIP), or RTPTap via Sangoma cards).<br>
<br>I have done this many times to catch intermittent problems that are continuously reported by users but cannot be readily reproduced. I just ask that the user log the time of the call and what they experienced, then I can listen to the recording, ascertain all the critical info that users leave off trouble reports, and figure out the commonalities. Obviously, all due notice/permission and/or legal disclosures should be made/given before recording anything.<br>
</div></div><br>It is great for troubleshooting (and yes, calls do get crossed and all kinds of other strangness in Asterisk, you know, what you write off as user error :-)<br><font color="#888888"><br>-- <br>Thanks,<br>Steve Totaro <br>
+18887771888 (Toll Free)<br>+12409381212 (Cell)<br></font><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>
<br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br>