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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 18th, 2012, 9:39 a.m., <b>Joshua Colp</b> wrote:</p>
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    <a href="https://reviewboard.asterisk.org/r/2165/diff/1/?file=31889#file31889line19238" style="color: black; font-weight: bold; text-decoration: underline;">branches/1.8/channels/chan_sip.c</a>
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     (Diff revision 1)

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   <td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static void handle_request_info(struct sip_pvt *p, struct sip_request *req)</pre></td>

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    <td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>
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    <td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">                        if ((ast_test_flag(&amp;p-&gt;flags[0], SIP_DTMF) == SIP_DTMF_INBAND)) {</pre></td>
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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;">It would be better if this tested if the value was SIP_DTMF_INFO or not, that way if RFC2833+INFO was received it would not duplicate DTMF in that instance either.</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Oh, and don&#39;t forget to update the log message.</pre>
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<p>- Joshua</p>


<br />
<p>On October 17th, 2012, 2:19 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. 17, 2012, 2:19 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>


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<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>

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




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