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      This is an automatically generated e-mail. To reply, visit:
      <a href="https://reviewboard.asterisk.org/r/2165/">https://reviewboard.asterisk.org/r/2165/</a>
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<blockquote style="margin-left: 1em; border-left: 2px solid #d0d0d0; padding-left: 10px;">
 <p style="margin-top: 0;">On October 19th, 2012, 8:19 a.m., <b>jcolp</b> wrote:</p>
 <blockquote style="margin-left: 1em; border-left: 2px solid #d0d0d0; padding-left: 10px;">
  <pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Per your comments about testing - I&#39;m not comfortable with INFO no longer working with auto, that&#39;s a pretty big behavior change. In that scenario we may have to have it continue to be broken for the DTMF received by multiple means situation. That&#39;s a much smaller group of people than those who use auto and INFO and have it just work right now.</pre>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Regarding &#39;auto&#39; and &#39;INFO&#39; on it&#39;s own not working, I agree it should work.
I also need to test &#39;auto&#39; and &#39;RFC2833&#39; on it&#39;s own.

I had a line in my previous comments, but got deleted when I edited and repasted my test results.
&#39;More thought required&#39;
</pre>
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<p>- Alec</p>


<br />
<p>On October 19th, 2012, 3:25 a.m., Alec Davis wrote:</p>






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<div>Review request for Asterisk Developers.</div>
<div>By Alec Davis.</div>


<p style="color: grey;"><i>Updated Oct. 19, 2012, 3:25 a.m.</i></p>




<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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   <pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Asterisk 1.8.16.0

file:/var/log/asterisk/dtmf when only &#39;2&#39; was hit once
[2012-10-17 19:46:17.879406] DTMF[17084] channel.c: DTMF begin &#39;2&#39; received on SIP/822-00000000
[2012-10-17 19:46:17.879467] DTMF[17084] channel.c: DTMF begin ignored &#39;2&#39; on SIP/822-00000000
[2012-10-17 19:46:17.950953] DTMF[17084] channel.c: DTMF end &#39;2&#39; received on SIP/822-00000000, duration 800 ms
[2012-10-17 19:46:17.951004] DTMF[17084] channel.c: DTMF end passthrough &#39;2&#39; on SIP/822-00000000
[2012-10-17 19:46:18.019135] DTMF[17084] channel.c: DTMF end &#39;2&#39; received on SIP/822-00000000, duration 51 ms
[2012-10-17 19:46:18.019228] DTMF[17084] channel.c: DTMF end passthrough &#39;2&#39; on SIP/822-00000000


In ASTERISK-20218 the attached file it can be seen that both PA2P.rtf has both &#39;inband&#39; and &#39;SIP INFO&#39; set. </pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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   <pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Asterisk SVN-branch-1.8-r375111M

I was able to verify the same conditions on a Grandstream GXP2000.

Below is after proposed patch:

file:/var/log/asterisk/dtmf when &#39;820&#39; was entered

[2012-10-17 20:16:18.883821] DTMF[23460] channel.c: DTMF begin &#39;8&#39; received on SIP/822-00000004
[2012-10-17 20:16:18.883868] DTMF[23460] channel.c: DTMF begin ignored &#39;8&#39; on SIP/822-00000004
[2012-10-17 20:16:18.963537] DTMF[23460] channel.c: DTMF end &#39;8&#39; received on SIP/822-00000004, duration 89 ms
[2012-10-17 20:16:18.963559] DTMF[23460] channel.c: DTMF end passthrough &#39;8&#39; on SIP/822-00000004
[2012-10-17 20:16:19.263805] DTMF[23460] channel.c: DTMF begin &#39;2&#39; received on SIP/822-00000004
[2012-10-17 20:16:19.263851] DTMF[23460] channel.c: DTMF begin ignored &#39;2&#39; on SIP/822-00000004
[2012-10-17 20:16:19.363423] DTMF[23460] channel.c: DTMF end &#39;2&#39; received on SIP/822-00000004, duration 89 ms
[2012-10-17 20:16:19.363444] DTMF[23460] channel.c: DTMF end passthrough &#39;2&#39; on SIP/822-00000004
[2012-10-17 20:16:19.643830] DTMF[23460] channel.c: DTMF begin &#39;0&#39; received on SIP/822-00000004
[2012-10-17 20:16:19.643876] DTMF[23460] channel.c: DTMF begin ignored &#39;0&#39; on SIP/822-00000004
[2012-10-17 20:16:19.783561] DTMF[23460] channel.c: DTMF end &#39;0&#39; received on SIP/822-00000004, duration 140 ms
[2012-10-17 20:16:19.783582] DTMF[23460] channel.c: DTMF end passthrough &#39;0&#39; on SIP/822-00000004


Console:

    -- Executing [s@voicemail-main:2] VoiceMailMain(&quot;SIP/822-00000004&quot;, &quot;&quot;) in new stack
    -- &lt;SIP/822-00000004&gt; Playing &#39;vm-login.gsm&#39; (language &#39;en&#39;)
[2012-10-17 20:16:18.935436] WARNING[23399]: chan_sip.c:19239 handle_request_info: Ignoring DTMF_INFO message as DTMF_INBAND is set on channel SIP/822-00000004
[2012-10-17 20:16:19.325618] WARNING[23399]: chan_sip.c:19239 handle_request_info: Ignoring DTMF_INFO message as DTMF_INBAND is set on channel SIP/822-00000004
[2012-10-17 20:16:19.734910] WARNING[23399]: chan_sip.c:19239 handle_request_info: Ignoring DTMF_INFO message as DTMF_INBAND is set on channel SIP/822-00000004
    -- &lt;SIP/822-00000004&gt; Playing &#39;vm-password.gsm&#39; (language &#39;en&#39;)
asterix*CLI&gt;
</pre>
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<div style="margin-top: 1.5em;">
 <b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>


 <a href="https://issues.asterisk.org/jira/browse/ASTERISK-20218">ASTERISK-20218</a>


</div>


<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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 <li>branches/1.8/channels/chan_sip.c <span style="color: grey">(375136)</span></li>

</ul>

<p><a href="https://reviewboard.asterisk.org/r/2165/diff/" style="margin-left: 3em;">View Diff</a></p>




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