Remco<div><br></div><div>Use hangup(3) and Asterisk will send a 404 </div><div><br></div><div>Excerpt from <a href="http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause">http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause</a>.</div>
<div><br></div><div><span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">ISUP Cause value SIP response</span></div><div><span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)"><br>
</span></div><div><span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">1 unallocated number 404 Not Found</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">2 no route to network 404 Not found</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">3 no route to destination 404 Not found</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">16 normal call clearing --- (*)</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">17 user busy 486 Busy here</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">18 no user responding 408 Request Timeout</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">19 no answer from the user 480 Temporarily unavailable</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">20 subscriber absent 480 Temporarily unavailable</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">21 call rejected 403 Forbidden (+)</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">22 number changed (w/o diagnostic) 410 Gone</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">22 number changed (w/ diagnostic) 301 Moved Permanently</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">23 redirection to new destination 410 Gone</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">26 non-selected user clearing 404 Not Found (=)</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">27 destination out of order 502 Bad Gateway</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">28 address incomplete 484 Address incomplete</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">29 facility rejected 501 Not implemented</span><br style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px">
<span style="font-family:Verdana,Geneva,Arial,Helvetica,sans-serif;font-size:12px;background-color:rgb(245,245,245)">31 normal unspecified 480 Temporarily unavailable</span><br clear="all"><div><br></div><div>Flavio E. Goncalves</div>
<div>SipPulse Routng and Billing Solutions for SIP. </div><div> </div><br>
<br><br><div class="gmail_quote">2012/10/14 Remco . <span dir="ltr"><<a href="mailto:remconl87@gmail.com" target="_blank">remconl87@gmail.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Thanks Max. That does the trick for the Asterisk part. However, calls are now returned with 603-declined. Anyone on how to make opensips wait for the message to complete, and then return 404?<br><br><div class="gmail_quote">
<div><div class="h5">
On Sat, Oct 13, 2012 at 11:06 PM, Max Mühlbronner <span dir="ltr"><<a href="mailto:mm@42com.com" target="_blank">mm@42com.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div class="h5">
<div link="blue" vlink="purple" lang="DE"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Hi,<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d" lang="EN-US">regarding asterisk as media-server, you could use the “noanswer” option for playback(). Then it will signal audio via progress messages but will not answer (200 OK) the call.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d" lang="EN-US"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d" lang="EN-US">Best Regards<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d" lang="EN-US"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d" lang="EN-US">Max M.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d" lang="EN-US"><u></u> <u></u></span></p><div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0cm 0cm 0cm">
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">Von:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a> [mailto:<a href="mailto:users-bounces@lists.opensips.org" target="_blank">users-bounces@lists.opensips.org</a>] <b>Im Auftrag von </b>Remco .<br>
<b>Gesendet:</b> Samstag, 13. Oktober 2012 22:52<br><b>An:</b> OpenSIPS users mailling list<br><b>Betreff:</b> [OpenSIPS-Users] Route to media-server, but reply negative<u></u><u></u></span></p></div><div><div>
<p class="MsoNormal"><u></u> <u></u></p><p class="MsoNormal">Hi all,<br><br>I have the following in the failure_route, for invalid destinations:<br><br> if(t_check_status("404")) {<br> # Dialed phone number does not exist<br>
<br> # Cancel call billing<br> resetflag(1);<br><br> # Start announcement<br> seturi("<a>sip:AN_invalidnumber@[ip of mediaserver]:5060</a>");<br>
t_relay();<br><br> #t_reply("404", "Not found");<br> exit;<br> }<br><br>When a 404 reply is received from upstream carrier(s) I would like to play an announcement to let the user know they made a mistake in the phone number.<br>
On the other hand, I would like those calls to show up as '404' in my statistics. Ideally the announcement should be played in the early media (don't know if that's possible with Asterisk as a media server?).<br>
The announcement works, however returns 200-OK. If I uncomment the 't_reply', the call is ended to soon without allowing the announcement to be played.<br><br>Does anyone how to solve this? I tried branching but I cannot get it to work.<br>
<br>Thanks,<br>Remco.<u></u><u></u></p></div></div></div></div><br></div></div>_______________________________________________<br>
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