Try to Add h extensions in frompstn context and print ${HANGUPCAUSE} in that you will receive in that ,<br><br>also read this for better implementation.<br><br><a href="http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause">http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause</a><br>
<br>regards<br>Dhaval<br><br><div class="gmail_quote">On Fri, Jul 29, 2011 at 11:58 AM, Nikhil <span dir="ltr">&lt;<a href="mailto:d.nikhil@cem-solutions.net">d.nikhil@cem-solutions.net</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<u></u>

  
    
    
  
  <div text="#000000" bgcolor="#ffffff">
    find the inline comment...<div class="im"><br>
    <br>
    On 07/29/2011 12:11 AM, Ishwar Sridharan wrote:
    <blockquote type="cite">The dialplan is very simple. When the call comes in,
      we hand the call over to adhearsion.<br>
      This is how the dialplan looks:<br>
      <br>
      ;group 0 will be used for incoming calls<br>
      EXOIN = DAHDI/g0<br>
      <br>
      ;group 11 for outgoing<br>
      EXOOUT = DAHDI/G11<br>
      <br>
      ;This will be used by adhearsion<br>
      EXOCID=xxxxxxxx<br>
      <br>
      [general]<br>
      autofallthrough = yes ;really?<br>
      clearglobalvars = no<br>
      <br>
      [frompstn]<br>
      ;Send everything to adhearsion<br>
      exten =&gt; _X.,1,Ringing<br>
      exten =&gt; _X.,n,AGI(agi://<a href="http://127.0.0.1" target="_blank">127.0.0.1</a>)<br>
    </blockquote></div>
     <font color="#ff6666">   exten =&gt; _X.,n,Hangup() ; Please try
      this.</font><div><div></div><div class="h5"><br>
    <blockquote type="cite"><br>
      ; End dialplan<br>
      <br>
      The rest of the logic happens in adhearsion.<br>
      <br>
      --<br>
      Thanks,<br>
      Ishwar.<br>
      <br>
      <br>
      <div class="gmail_quote">On Thu, Jul 28, 2011 at 6:33 PM, Nikhil <span dir="ltr">&lt;<a href="mailto:d.nikhil@cem-solutions.net" target="_blank">d.nikhil@cem-solutions.net</a>&gt;</span>
        wrote:<br>
        <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
          <div text="#000000" bgcolor="#ffffff"> Can you share the
            dialplan ,where SIP call is dialing...<br>
            Thanks<br>
            Nikhil
            <div>
              <div><br>
                <br>
                On 07/28/2011 06:15 PM, Ishwar Sridharan wrote: </div>
            </div>
            <blockquote type="cite">
              <div>
                <div>Hello everybody,<br>
                  <br>
                  We have an asterisk 1.8.4.1 setup, connected to a PRI
                  line.<br>
                  <br>
                  We&#39;re currently facing an issue where asterisk does
                  not recognise the event when the called party
                  declines/cuts the call. This happens specifically over
                  calls on a PRI line. For calls over SIP, call decline
                  event is captured properly.<br>
                  <br>
                  I wasn&#39;t able to find a solution on the asterisk-users
                  mailing list archive. Any suggestions/help would be
                  much appreiciated :) I can share the relevant parts of
                  the configuration files, if needed.<br>
                  <br>
                  Here&#39;s an excerpt from asterisk logs for a SIP call.<br>
                      -- SIP/xxxxx-00000000 requested special control
                  16, passing it to SIP/xxxxx-00000001<br>
                      -- Started music on hold, class &#39;default&#39;, on
                  SIP/xxxxx-00000001<br>
                      -- SIP/xxxxx-00000000 requested special control
                  20, passing it to SIP/xxxxx-00000001<br>
                      -- Got SIP response 603 &quot;Decline&quot; back from <a href="http://127.0.0.1:5063/" target="_blank">127.0.0.1:5063</a><br>
                      -- SIP/xxxxx-00000001 is busy<br>
                      -- Stopped music on hold on SIP/xxxxx-00000001<br>
                  <br>
                  As you can see, on a SIP call, a call reject event is
                  identified.<br>
                  <br>
                  For a call over the PRI, on the other hand, this event
                  is not recognised. Here&#39;s an excerpt from asterisk log
                  for a call over PRI.<br>
                  Call from yyyy to xxxx.<br>
                      -- Requested transfer capability: 0x10 - 3K1AUDIO<br>
                      -- Called G11/xxxxx<br>
                      -- Started music on hold, class &#39;default&#39;, on
                  DAHDI/i1/yyyyy<br>
                      -- DAHDI/i1/xxxxx-18f8 is proceeding passing it to
                  DAHDI/i1/yyyyy<br>
                      -- DAHDI/i1/xxxxx-18f8 is ringing<br>
                  # At this point in time, xxxxx rejects the call. The
                  event that&#39;s logged in asterisk is the following:<br>
                      -- DAHDI/i1/xxxxx-18f8 is making progress passing
                  it to DAHDI/i1/yyyyy<br>
                  # And the call times out after the default 30s.<br>
                      -- Nobody picked up in 30000 ms<br>
                  <br>
                  Is there a reason why asterisk doesn&#39;t recognise the
                  &quot;call decline&quot;, and does it need any configuration
                  changes to enable this?<br>
                  <br>
                  Thanks for your help.<br>
                  <br>
                  --<br>
                  Cheers,<br>
                  <font color="#888888">Ishwar.</font><br>
                </div>
              </div>
              <pre><fieldset></fieldset>
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            </blockquote>
            <br>
          </div>
          <br>
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        </blockquote>
      </div>
      <br>
      <pre><fieldset></fieldset>
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    </blockquote>
    <br>
  </div></div></div>

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<br>
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