<div dir="ltr"><div><div>Peter,<br><br>FYI, I tested this patch and it worked for me. I suspect that you need to use export instead of set for the variables. For testing I simply used the Local_Extension in the default.xml file. I added these two lines directly before the voicemail action, so it looks like this:<br>
<br>        &lt;action application=&quot;export&quot; data=&quot;skip_record_urgent_check=true&quot;/&gt;<br>        &lt;action application=&quot;export&quot; data=&quot;skip_record_check=true&quot;/&gt;<br>        &lt;action application=&quot;bridge&quot; data=&quot;loopback/app=voicemail:default ${domain_name} ${dialed_extension}&quot;/&gt;<br>
<br></div>The message gets recorded and then the caller simply hears &quot;goodbye&quot; and gets disconnected.<br></div>-MC<br><br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Apr 29, 2013 at 8:18 AM, Michael Collins <span dir="ltr">&lt;<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Thanks for testing. If I get a moment to work on this today I will. It&#39;s probably something simple.<span class="HOEnZb"><font color="#888888"><br>
</font></span></div><span class="HOEnZb"><font color="#888888">-MC<br></font></span></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Apr 29, 2013 at 4:15 AM, Peter Steinbach <span dir="ltr">&lt;<a href="mailto:lists@telefaks.de" target="_blank">lists@telefaks.de</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
  
    
  
  <div bgcolor="#FFFFFF" text="#000000">
    <div>Hello Michael,<br>
      <br>
      thanks for your quick support. <br>
      I applied the patch and tested it, however, it did not suppress
      the messages:<br>
      Here&#39;s an excerpt from the log with the relevant parts marked in
      bold:<br>
      <br>
      <b>EXECUTE <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a>
        set(skip_record_check=true)</b><b><br>
      </b><b>2013-04-29 13:07:51.694194 [DEBUG] mod_dptools.c:1344
        <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> SET [skip_record_check]=[true]</b><b><br>
      </b><b>EXECUTE <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a>
        set(skip_urgent_check=true)</b><b><br>
      </b><b>2013-04-29 13:07:51.694194 [DEBUG] mod_dptools.c:1344
        <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> SET [skip_urgent_check]=[true]</b><b><br>
      </b>EXECUTE <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> answer()<br>
      2013-04-29 13:07:51.704184 [DEBUG] sofia_glue.c:3351 AUDIO RTP
      [<a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a>] 192.168.178.220 port 12522 -&gt;
      192.168.178.118 port 3000 codec: 8 ms: 20<br>
      2013-04-29 13:07:51.704184 [DEBUG] switch_rtp.c:1975 Starting
      timer [soft] 160 bytes per 20ms<br>
      2013-04-29 13:07:51.704184 [DEBUG] switch_rtp.c:2331 RTCP send
      rate is: 5000 and packet rate is: 20000 Remote Port: 3001<br>
      2013-04-29 13:07:51.704184 [DEBUG] switch_rtp.c:1342 Setting RTCP
      remote addr to <a href="http://192.168.178.118:3001" target="_blank">192.168.178.118:3001</a><br>
      2013-04-29 13:07:51.704184 [DEBUG] sofia_glue.c:3615 Set 2833 dtmf
      send payload to 101<br>
      2013-04-29 13:07:51.704184 [DEBUG] sofia_glue.c:3621 Set 2833 dtmf
      receive payload to 101<br>
      2013-04-29 13:07:51.704184 [DEBUG] sofia_glue.c:3648
      <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> Set rtp dtmf delay to 40<br>
      2013-04-29 13:07:51.704184 [DEBUG] mod_sofia.c:856 Local SDP
      <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a>:<br>
      v=0<br>
      o=FreeSWITCH 1367221149 1367221150 IN IP4 192.168.178.220<br>
      s=FreeSWITCH<br>
      c=IN IP4 192.168.178.220<br>
      t=0 0<br>
      m=audio 12522 RTP/AVP 8 101<br>
      a=rtpmap:8 PCMA/8000<br>
      a=rtpmap:101 telephone-event/8000<br>
      a=fmtp:101 0-16<br>
      a=silenceSupp:off - - - -<br>
      a=ptime:20<br>
      a=sendrecv<br>
      <br>
      2013-04-29 13:07:51.704184 [DEBUG] switch_core_session.c:830 Send
      signal <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> [BREAK]<br>
      2013-04-29 13:07:51.704184 [DEBUG] switch_channel.c:3395
      (<a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a>) Callstate Change RINGING -&gt;
      ACTIVE<br>
      2013-04-29 13:07:51.704184 [NOTICE] mod_dptools.c:1176 Channel
      [<a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a>] has been answered<br>
      2013-04-29 13:07:51.704184 [DEBUG] switch_core_session.c:975 Send
      signal <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> [BREAK]<br>
      2013-04-29 13:07:51.704184 [DEBUG] sofia.c:5574 Channel
      <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> entering state [completed][200]<br>
      <b>EXECUTE <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> voicemail(default
        <a href="http://mydomain.com" target="_blank">mydomain.com</a> 200)</b><br>
      2013-04-29 13:07:51.724183 [DEBUG] switch_core_session.c:975 Send
      signal <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> [BREAK]<br>
      2013-04-29 13:07:51.724183 [DEBUG] switch_core_session.c:975 Send
      signal <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> [BREAK]<br>
      2013-04-29 13:07:51.724183 [DEBUG] switch_core_session.c:975 Send
      signal <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> [BREAK]<br>
      2013-04-29 13:07:51.734185 [DEBUG] sofia.c:5574 Channel
      <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> entering state [ready][200]<br>
      2013-04-29 13:07:51.734185 [DEBUG] switch_ivr.c:192 Codec
      Activated L16@8000hz 1 channels 20ms<br>
      2013-04-29 13:07:51.844184 [DEBUG] switch_ivr_play_say.c:1338
      Codec Activated L16@8000hz 1 channels 20ms<br>
      2013-04-29 13:07:51.884186 [DEBUG] switch_rtp.c:3694 Correct
      ip/port confirmed.<br>
      2013-04-29 13:07:58.204183 [DEBUG] switch_ivr_play_say.c:1721 done
      playing file
/usr/local/freeswitch/storage/voicemail/default/<a href="http://mydomain.com/200/greeting_1.wav" target="_blank">mydomain.com/200/greeting_1.wav</a><br>
      2013-04-29 13:07:59.224199 [DEBUG] switch_ivr_play_say.c:595 Raw
      Codec Activated<br>
      2013-04-29 13:07:59.224199 [DEBUG] switch_core_codec.c:219
      <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> Push codec L16:70<br>
      2013-04-29 13:08:07.264182 [DEBUG] switch_core_codec.c:244
      <a href="mailto:sofia/internal/208@mydomain.com" target="_blank">sofia/internal/208@mydomain.com</a> Restore previous codec PCMA:8.<br>
      2013-04-29 13:08:07.264182 [DEBUG] switch_ivr_play_say.c:70 No
      language specified - Using [en]<br>
      2013-04-29 13:08:07.264182 [WARNING] switch_xml.c:2356 Invalid
      UTF-8 character to ampersand, skip it<br>
      2013-04-29 13:08:07.264182 [WARNING] switch_xml.c:2356 Invalid
      UTF-8 character to ampersand, skip it<br>
      2013-04-29 13:08:07.264182 [WARNING] switch_xml.c:2356 Invalid
      UTF-8 character to ampersand, skip it<br>
      2013-04-29 13:08:07.264182 [WARNING] switch_xml.c:2356 Invalid
      UTF-8 character to ampersand, skip it<br>
      <b>2013-04-29 13:08:07.264182 [DEBUG] switch_ivr_play_say.c:247
        Handle play-file:[voicemail/vm-press.wav] (en:en)</b><b><br>
      </b>2013-04-29 13:08:07.264182 [DEBUG] switch_ivr_play_say.c:1338
      Codec Activated L16@8000hz 1 channels 20ms<br>
      2013-04-29 13:08:07.664181 [DEBUG] switch_ivr_play_say.c:1721 done
      playing file
      /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-press.wav<br>
      <br>
      Best regards<span><font color="#888888"><br>
      Peter</font></span><div><div><br>
      On 04/29/13 07:15, Michael Collins wrote:<br>
    </div></div></div><div><div>
    <blockquote type="cite">
      <div dir="ltr">
        <div>
          <div>
            <div>
              <div>Peter,<br>
                <br>
              </div>
              <div>After looking at mod_voicemail.c I could tell that
                there&#39;s no way to do this currently. However, like Brian
                said, there isn&#39;t a whole lot to it. Here&#39;s a simple
                patch for you to test. <br>
              </div>
              <div><br>
              </div>
              To use this just set the channel variables
              &#39;skip_record_check&#39; and &#39;skip_urgent_check&#39; to true. It
              should then just disconnect the channel once the recording
              is done. The first variable turns off the record check (&quot;1
              to listen, 2 to save, 3 to re-record&quot;) and the second
              turns off the urgent check (&quot;to mark this message urgent,
              press 1...&quot;).<br>
              <br>
            </div>
            Also, if this seems to work then please open a new jira of
            type &quot;improvement&quot; for project &quot;FreeSWITCH&quot; and attach this
            patch. (If you make any adjustments, please create a new
            patch and attach it instead of this file.)<br>
            <br>
          </div>
          Let us know how it goes.<br>
        </div>
        -Michael<br>
        <br>
      </div>
      <div class="gmail_extra"><br>
        <br>
        <div class="gmail_quote">On Sun, Apr 28, 2013 at 3:30 AM, Peter
          Steinbach <span dir="ltr">&lt;<a href="mailto:lists@telefaks.de" target="_blank">lists@telefaks.de</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <div bgcolor="#FFFFFF" text="#000000">
              <div>Hello Brian,<br>
                <br>
                I was thinking about that too. But is there a way of
                having a solution without patching the code?<br>
                <br>
                <br>
                Best regards<span><font color="#888888"><br>
                    Peter</font></span>
                <div>
                  <div><br>
                    <br>
                    <br>
                    <br>
                    <br>
                    On 04/26/13 15:45, Brian West wrote:<br>
                  </div>
                </div>
              </div>
              <blockquote type="cite">
                <div>
                  <div>
                    <pre>Should be fairly simple to find where that ends in the code and add the proper bits so it doesn&#39;t play that.

Its patches just like what you described that got me into coding C back when I started with Asterisk.

On Apr 26, 2013, at 7:01 AM, Peter Steinbach <a href="mailto:lists@telefaks.de" target="_blank">&lt;lists@telefaks.de&gt;</a> wrote:

</pre>
                    <blockquote type="cite">
                      <pre>Anybody has a clue how to solve this?
</pre>
                    </blockquote>
                    <pre>--
Brian West
<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
T: <a href="tel:%2B1.918.420.9266" value="+19184209266" target="_blank">+1.918.420.9266</a>  |  F: <a href="tel:%2B1.918.420.9267" value="+19184209267" target="_blank">+1.918.420.9267</a>  |  M: +1.918.424.WEST
iNUM: <a href="tel:%2B883%205100%201420%209266" value="+883510014209266" target="_blank">+883 5100 1420 9266</a>
ISN: 410*543






</pre>
                    <br>
                    <fieldset></fieldset>
                    <br>
                  </div>
                </div>
                <div>
                  <pre>_________________________________________________________________________
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</pre>
                </div>
              </blockquote>
              <div> <br>
                <br>
                <pre cols="72">-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
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</pre>
              </div>
            </div>
            <br>
_________________________________________________________________________<br>
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            <br>
          </blockquote>
        </div>
        <br>
        <br clear="all">
        <br>
        -- <br>
        Michael S Collins<br>
        Twitter: @mercutioviz<br>
        <a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br>
        <a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br>
        <a href="http://www.OSTAG.org" target="_blank">http://www.OSTAG.org</a><br>
        <br>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      <pre>_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a>

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Official FreeSWITCH Sites
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</pre>
    </blockquote>
    <br>
    <br>
    <pre cols="72">-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
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</pre>
  </div></div></div>

<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
<a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<br></blockquote></div><br><br clear="all"><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br>

<a href="http://www.OSTAG.org" target="_blank">http://www.OSTAG.org</a><br><br>
</div>
</div></div></blockquote></div><br><br clear="all"><br>-- <br>Michael S Collins<br>Twitter: @mercutioviz<br><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br><a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br>
<a href="http://www.OSTAG.org" target="_blank">http://www.OSTAG.org</a><br><br>
</div>