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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, 3:24 p.m., <b>Joshua Colp</b> wrote:</p>
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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;">I think we should have *some* kind of message. If someone doesn't configure things properly they'll be confused very very fast when DTMF comes in via INFO and nothing happens.</pre>
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<p>On October 18th, 2012, 3:45 p.m., <b>Alec Davis</b> wrote:</p>
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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;">The reason why I agreed with opticron, if you're operating asterisk is a hosted PBX environment, where the clients have the INFO+INBAND or INFO+RFC2833 setting wrong already. A new release would spam the CLI, each time they press a digit if the warning was still there.
I'm also wondering now about other permutations;
dtmfmode=auto
What if the client sends RFC2833+INBAND+INFO.
There are 7 senarios to consider from a client point of view.
And a few less from Asterisk's dtmfmode point of view.</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;">Yes, and I'm thinking from a simple user perspective. If their configuration was/is incorrect we would now silently drop the DTMF and give no warning. That's not good. As for auto stuff can certainly fall apart with that, I personally think you should be explicit with DTMF.</pre>
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<p>- Joshua</p>
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<p>On October 18th, 2012, 3:21 p.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. 18, 2012, 3:21 p.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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