<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 14 (filtered medium)"><style><!--
/* Font Definitions */
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
{font-family:Tahoma;
panose-1:2 11 6 4 3 5 4 4 2 4;}
@font-face
{font-family:Consolas;
panose-1:2 11 6 9 2 2 4 3 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
pre
{mso-style-priority:99;
mso-style-link:"HTML Preformatted Char";
margin:0in;
margin-bottom:.0001pt;
font-size:10.0pt;
font-family:"Courier New";}
span.hoenzb
{mso-style-name:hoenzb;}
span.HTMLPreformattedChar
{mso-style-name:"HTML Preformatted Char";
mso-style-priority:99;
mso-style-link:"HTML Preformatted";
font-family:Consolas;}
span.EmailStyle20
{mso-style-type:personal-reply;
font-family:"Calibri","sans-serif";
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;
font-family:"Calibri","sans-serif";}
@page WordSection1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.WordSection1
{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Isn’t it easier just to use a SIP door phone?<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> --Don<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Ishfaq Malik<br><b>Sent:</b> Friday, December 20, 2013 10:31 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Reading DTMF sent by callee during a SIP call<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal>On 20 December 2013 16:13, Alex <<a href="mailto:ralienpp@gmail.com" target="_blank">ralienpp@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>Hi everyone,<br><br>I am looking for advice about the design of a SIP-based intercom. I<br>count on your help, as my current attempts are not fruitful (yet).<br><br>This will be a pretty long message, so here's my fundamental question:<br><br>Is there a way to interpret DTMF tones sent by the calee<br>(not the caller) while a voice call is in progress?<br><br><br><br><br><br><br>Here's the desired scenario:<br><br>- there is a box with speakers and a mic<br>- Asterisk is running on a computer inside that box<br>- the box is embedded in a door<br>- There are two user accounts, UserA and userB<br>- UserA is a client that runs on the server*<br>- UserA calls UserB and they are having a voice conversation<br><br><br>Throughout the call, Asterisk must react to DTMF tones sent by userB;<br>such that an action is executed when a specific key is pressed.<br><br>The idea is to build an intercom that would enable me to open a door<br>remotely, by relying entirely on SIP, so there would be no need to<br>have some additional communication channel to send the "open door"<br>signal.<br><br><br><br><br>I have previously implemented IVRs using `Background` and jumped to<br>specific extensions, when a button was pressed. But in that case, the<br>extensions are dialed by the caller; whereas now the input must from<br>the person who answered the call.<br><br>If I use `Dial` and `Read` - the latter is only executed after `Dial`<br>terminates - so this is not suitable.<br><br><br>`Background` behaves like I need - but it plays back a predefined<br>file, so it is not suitable for an interactive conversation.<br><br><br><br>* Having a SIP client on the same machine as the Asterisk server<br>itself is not possible, because both won't be able to bind to port<br>5060. My guess is that the solution is to originate a call from the<br>CLI; but I haven't gotten to that part yet.<br><br><br><br><br>Thank you for your patience, I am looking forward to your feedback,<br>Alex<br><br><o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>You could create your own feature in features.conf that executes a Macro/Gosub defined in sip.conf...<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Ish<o:p></o:p></p></div></div><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal>-- <o:p></o:p></p><div><pre>Ishfaq Malik <o:p></o:p></pre><pre>Department: VOIP Support<o:p></o:p></pre><pre>Company: Packnet Limited<o:p></o:p></pre><pre>t: +44 (0)845 004 4994<o:p></o:p></pre><pre>f: +44 (0)161 660 9825<o:p></o:p></pre><pre>e: <a href="mailto:ish@pack-net.co.uk" target="_blank">ish@pack-net.co.uk</a><o:p></o:p></pre><pre>w: <a href="http://www.pack-net.co.uk" target="_blank">http://www.pack-net.co.uk</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Registered Address: PACKNET LIMITED, Duplex 2, Ducie House<o:p></o:p></pre><pre>37 Ducie Street <o:p></o:p></pre><pre>Manchester, M1 2JW<o:p></o:p></pre><pre>COMPANY REG NO. 04920552<o:p></o:p></pre></div></div></div></div></body></html>