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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial Unicode MS","sans-serif";color:#3A3ED2'>Hangup cause gives ISDN cause codes. <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial Unicode MS","sans-serif";color:#3A3ED2'>Its easier with sip 4xx or 5xx as there are only 2 to check. So, looking to get sip reason.</span><span style='font-size:11.0pt;font-family:"Arial","sans-serif";color:#3A3ED2'><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial Unicode MS","sans-serif";color:#3A3ED2'><o:p> </o:p></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"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Robert Thomas<br><b>Sent:</b> 03 March 2011 02:28<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Failover Routing<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>What value do you get from the hangup cause, are they different?<o:p></o:p></p><div><p class=MsoNormal> <o:p></o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'>I think can you use a gotoif checking the hangup cause.<o:p></o:p></p><div><p class=MsoNormal>On Wed, Mar 2, 2011 at 12:43 PM, Tilghman Lesher <<a href="mailto:tilghman@meg.abyt.es">tilghman@meg.abyt.es</a>> wrote:<o:p></o:p></p><div><p class=MsoNormal style='margin-bottom:12.0pt'>On Wednesday 02 March 2011 07:06:31 Andrew Thomas wrote:<br>> It seems like it is a v1.8 only function at present (unless a backport<br>> is released).<br>><br>> From <a href="http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause" target="_blank">http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause</a><br>><br>> -----<br>> Asterisk 1.8 will allow to read SIP response codes in the dialplan via<br>><br>> ${HASH(SIP_CAUSE,<channel-name>)}<br>><br>> Asterisk 1.8 also comes with a 'use_q850_reason' configuration option<br>> for generating and parsing, if available: -----<br>><br>> That will give you what you want if you consider upgrading to v1.8.<o:p></o:p></p></div><p class=MsoNormal>A backport on this is not possible. It depends upon some core<br>functionality introduced in the 1.8 branch.<br><br>--<br>Tilghman<br><span style='color:#888888'><br>--</span><o:p></o:p></p><div><div><p class=MsoNormal>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</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><o:p></o:p></p></div></div></div><p class=MsoNormal><br><br clear=all><br>-- <br>Robert<o:p></o:p></p></div></div></body></html>