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<div><div><font class="Apple-style-span" face="Tahoma" size="2">Hello again,</font></div><div><font class="Apple-style-span" face="Tahoma" size="2"><br></font></div><div><font class="Apple-style-span" face="Tahoma" size="2">So I have made and analyse the two *.cap files. the first file contains call from pstn with dtmf issue. the second file contain call from sip softphone without dtmf issue.</font></div><div><font class="Apple-style-span" face="Tahoma" size="2"><br></font></div><div><font class="Apple-style-span" face="Tahoma" size="2">As far as I see when I call from softphone and send dtmf, in this moment the media stream is empty (because softphone cut dtmf from media in accordance with rfc2833).</font></div><div><font class="Apple-style-span" face="Tahoma" size="2"><br></font></div><div><font class="Apple-style-span" face="Tahoma" size="2">But when I call from pstn (from cell phone for example) and send dtmf, in this moment the media stream is not empty. I see the hits which <span class="Apple-tab-span" style="white-space:pre"> </span>correspond the sending of dtmf. I think that therefore I receive double dtmf on endpoint. </font></div><div><font class="Apple-style-span" face="Tahoma" size="2"><br></font></div><div><font class="Apple-style-span" face="Tahoma" size="2">How I can to suppress dtmf in inband ?</font></div><div><font class="Apple-style-span" face="Tahoma" size="2"><br></font></div><div><font class="Apple-style-span" face="Tahoma" size="2">"</font><span class="Apple-style-span" style="font-family: Tahoma; font-size: 13px; ">Please let me know if you have addressed this issue."</span></div><div><font class="Apple-style-span" face="Tahoma" size="2">Sure.</font></div><div><font class="Apple-style-span" face="Tahoma" size="2"><br></font></div><div><font class="Apple-style-span" face="Tahoma" size="2">Thanks.</font></div></div><div style="font-family: Tahoma; font-size: 10pt; "><br></div><div style="font-family: Tahoma; font-size: 10pt; "><hr id="stopSpelling">Date: Wed, 2 Nov 2011 10:44:08 +0530<br>From: vellayappan.n@mobax.com<br>To: freeswitch-users@lists.freeswitch.org<br>Subject: Re: [Freeswitch-users] double dtmf<br><br>Hi,<br><br>I too have the same issue. When calling from PSTN number, the freeswitch says Incoming call from "0000000000" instead of the actual number. Moreover it detects only few digits (say 2 or 3) not all the dtmf numbers. <br>
<br>Please let me know if you have addressed this issue.<br><br> <br><br><div class="ecxgmail_quote">2011/11/2 Yuriy Nasida <span dir="ltr"><<a href="mailto:nasida@live.ru">nasida@live.ru</a>></span><br><blockquote class="ecxgmail_quote" style="padding-left:1ex">
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<div><font face="Tahoma" size="2">Thank you for your answer.</font></div><div class="ecxim"><div><font face="Tahoma" size="2"><br></font></div><div><font face="Tahoma" size="2">"are you doing something explicitly to send the DTMF"</font></div>
</div><div><font face="Tahoma" size="2">No. I just press buttons in my cell phone (I mean startpoint). </font></div><div><font face="Tahoma" size="2"><br></font></div><div><font face="Tahoma" size="2">In my opinion it is strange. I make answer() in lua script and use session:getDigits(). All works fine.</font></div>
<div><font face="Tahoma" size="2">In the end of script I use execute_extension which implements bridge.</font></div><div><font face="Tahoma" size="2">session:execute("execute_extension", "".. recieved_digit .." XML ".. user_context .."")</font></div>
<div><font face="Tahoma" size="2"><br></font></div><div><font face="Tahoma" size="2">Thus the call was answered in lua script. Next the call go to endpoint. How dtmf must be transmitted in this case ?</font></div><div><font face="Tahoma" size="2"><br>
</font></div><div><font face="Tahoma" size="2">Why I have not issues when I call from sip devices?</font></div><div><font face="Tahoma" size="2"><br></font></div><div><font face="Tahoma" size="2">Any help are welcomed.</font></div>
<div><font face="Tahoma" size="2"><br></font></div><div><font face="Tahoma" size="2">Thanks.</font></div><div><font face="Tahoma" size="2"><br></font></div><div style="font-family:Tahoma;font-size:10pt"><hr>From: <a href="mailto:avi@avimarcus.net">avi@avimarcus.net</a><br>
Date: Tue, 1 Nov 2011 21:22:11 +0200<br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>Subject: Re: [Freeswitch-users] double dtmf<div><div></div><div class="h5">
<br><br><div dir="ltr">"<span style="font-family:Tahoma;background-color:rgb(255, 255, 255)">when I try to send dtmf second time for endpoint " - are you doing something explicitly to send the DTMF?</span><div>
<font face="Tahoma">I just ignore the dtmf, and when FS bridges it sends the call on properly.</font></div><div><font face="Tahoma"> <br clear="all"></font><div dir="ltr">
<span style="font-family:Verdana,Arial,Helvetica,sans-serif"><span style="font-size:small">-Avi</span></span></div>
<br><br><div>2011/11/1 Yuriy Nasida <span dir="ltr"><<a href="mailto:nasida@live.ru">nasida@live.ru</a>></span><br><blockquote style="padding-left:1ex">
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<div><font face="Tahoma" size="2">Hello community,</font></div><div><font face="Tahoma" size="2"><br></font></div><div><font face="Tahoma" size="2">I have some issue with delivery of dtmf.</font></div><div><font face="Tahoma" size="2">The call flow is:</font></div>
<div><font face="Tahoma" size="2">one sip provider - FS - second sip provider - endpoint which expects correct dtmf</font></div><div><font face="Tahoma" size="2"><br></font></div><div><font face="Tahoma" size="2">On FS I have simple custom lua application which implement logic of calling cards system.</font></div>
<div><font face="Tahoma" size="2"><br></font></div><div><font face="Tahoma" size="2">I have not issues when I call to lua application. I can enter destination number (by using dtmf apparently) and all works fine. But when I try to send dtmf second time for endpoint which expects correct dtmf I have issue with double dtmf. </font></div>
<div><font face="Tahoma" size="2"><br></font></div><div><font face="Tahoma" size="2">It is noted that when I call from sip device I have not ANY issues. But I have issue when I call from any cell phone or landline phone.</font></div>
<div><font face="Tahoma" size="2"><br></font></div><div><font face="Tahoma" size="2">Any ideas are welcomed.</font></div><div><font face="Tahoma" size="2"><br></font></div><div><font face="Tahoma" size="2">Thanks.</font></div>
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