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<p>Asterisk is in version 14.7.1. One end is a SIP Trunk to another
Asterisk, the other end a home-made SIP phone. SIP INFO requests
are coming from the other Asterisk.</p>
<p>Both endpoints use chan_sip with "dtmfmode" set to "info".</p>
<p>This is not strictly speaking a one-to-one setup since we're
connecting to a SIP Trunk which then connects to another SIP
phone, but I think it doesn't make much difference regarding SIP
INFO handling.<br>
</p>
<br>
<div class="moz-cite-prefix">Le 15/12/2017 à 12:12, Olivier a
écrit :<br>
</div>
<blockquote type="cite"
cite="mid:CAPeT9jieVC+sL1vogoQS1L9wRqenrCesMJqmK_qHrBa0NFdbXw@mail.gmail.com">
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<div>Hello Jean,<br>
<br>
</div>
1. Can you describe a bit further how both ends of the above
call were both made of and configured ?<br>
</div>
<div>DTMF receiving is Asterisk/SIP channel but which version
?<br>
</div>
<div>Is the other end a SIP phone or a SIP trunk ?<br>
</div>
<div><br>
</div>
2. Do you observe such behaviour in a one-to-one setup (one
end emits, the other listen) or does the DTMF sending side
also communicates with an other endpoint ?<br>
<br>
</div>
Cheers<br>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2017-12-13 12:22 GMT+01:00 Jean Aunis <span
dir="ltr"><<a href="mailto:jean.aunis@prescom.fr"
target="_blank" moz-do-not-send="true">jean.aunis@prescom.fr</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
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<p>Hello,</p>
<p>I think there is an issue when DTMF are handled with
SIP INFO and direct media is enabled.</p>
<p>When I receive a SIP INFO, the logs tell me that a
"DTMF begin" is generated, but no related "DTMF end" is
generated, unless the call is ended. Here is an excerpt
of the logs :</p>
<p><b><tt>--- SIP INFO received </tt></b><b><tt>on</tt> </b><b><tt>SIP/xxx-00000004:</tt></b></p>
<p><tt>[Dec 13 11:56:16] DTMF[18193][C-00000005]
channel.c: DTMF end '#' received on SIP/xxx-00000004,
duration 257 ms</tt><tt><br>
</tt><tt>[Dec 13 11:56:16] DTMF[18193][C-00000005]
channel.c: DTMF begin emulation of '#' with duration
257 queued on SIP/xxx-00000004</tt></p>
<p><b><tt>--- </tt></b><tt><tt><b>SIP/xxx-00000004 </b><b>is
hanged up:</b><br>
</tt></tt></p>
<p><tt>[Dec 13 11:56:19] VERBOSE[18193][C-00000005]
bridge_channel.c: Channel SIP/xxx-00000004 left
'native_rtp' basic-bridge <4a5905ac-29f8-41c5-9981-<wbr>e9d0f4966c56></tt><tt><br>
</tt><tt>[Dec 13 11:56:19] DTMF[18193][C-00000005]
bridge_channel.c: DTMF end '#' simulated to bridge
4a5905ac-29f8-41c5-9981-<wbr>e9d0f4966c56 because
SIP/xxx-00000004 left. Duration 3012 ms.</tt></p>
<p>Do you think it is a bug ? I would tend to say yes, but
I'm not so sure.</p>
<p>Regards</p>
<span class="HOEnZb"><font color="#888888">
<p>Jean Aunis<br>
</p>
</font></span></div>
<br>
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