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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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<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(&p->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't forget to update the log message.</pre>
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<p>- Joshua</p>
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<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 '2' was hit once
[2012-10-17 19:46:17.879406] DTMF[17084] channel.c: DTMF begin '2' received on SIP/822-00000000
[2012-10-17 19:46:17.879467] DTMF[17084] channel.c: DTMF begin ignored '2' on SIP/822-00000000
[2012-10-17 19:46:17.950953] DTMF[17084] channel.c: DTMF end '2' received on SIP/822-00000000, duration 800 ms
[2012-10-17 19:46:17.951004] DTMF[17084] channel.c: DTMF end passthrough '2' on SIP/822-00000000
[2012-10-17 19:46:18.019135] DTMF[17084] channel.c: DTMF end '2' received on SIP/822-00000000, duration 51 ms
[2012-10-17 19:46:18.019228] DTMF[17084] channel.c: DTMF end passthrough '2' on SIP/822-00000000
In ASTERISK-20218 the attached file it can be seen that both PA2P.rtf has both 'inband' and 'SIP INFO' 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 '820' was entered
[2012-10-17 20:16:18.883821] DTMF[23460] channel.c: DTMF begin '8' received on SIP/822-00000004
[2012-10-17 20:16:18.883868] DTMF[23460] channel.c: DTMF begin ignored '8' on SIP/822-00000004
[2012-10-17 20:16:18.963537] DTMF[23460] channel.c: DTMF end '8' received on SIP/822-00000004, duration 89 ms
[2012-10-17 20:16:18.963559] DTMF[23460] channel.c: DTMF end passthrough '8' on SIP/822-00000004
[2012-10-17 20:16:19.263805] DTMF[23460] channel.c: DTMF begin '2' received on SIP/822-00000004
[2012-10-17 20:16:19.263851] DTMF[23460] channel.c: DTMF begin ignored '2' on SIP/822-00000004
[2012-10-17 20:16:19.363423] DTMF[23460] channel.c: DTMF end '2' received on SIP/822-00000004, duration 89 ms
[2012-10-17 20:16:19.363444] DTMF[23460] channel.c: DTMF end passthrough '2' on SIP/822-00000004
[2012-10-17 20:16:19.643830] DTMF[23460] channel.c: DTMF begin '0' received on SIP/822-00000004
[2012-10-17 20:16:19.643876] DTMF[23460] channel.c: DTMF begin ignored '0' on SIP/822-00000004
[2012-10-17 20:16:19.783561] DTMF[23460] channel.c: DTMF end '0' received on SIP/822-00000004, duration 140 ms
[2012-10-17 20:16:19.783582] DTMF[23460] channel.c: DTMF end passthrough '0' on SIP/822-00000004
Console:
-- Executing [s@voicemail-main:2] VoiceMailMain("SIP/822-00000004", "") in new stack
-- <SIP/822-00000004> Playing 'vm-login.gsm' (language 'en')
[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
-- <SIP/822-00000004> Playing 'vm-password.gsm' (language 'en')
asterix*CLI>
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<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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