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<div class="moz-cite-prefix">On 28/11/13 15:36, Salaheddine Elharit
wrote:<br>
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<blockquote
cite="mid:CAHexamvWTn2tZULtMOcteBLRa5wngH0yN49qBqNMQmLFybbc3Q@mail.gmail.com"
type="cite">
<div dir="ltr">hi
<div style="">i follow your dialplan but the issue still the
same ican't stop the speech and go to another context </div>
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<div style="">any other idea please <br>
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<div style=""><br>
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<div style="">best regards .</div>
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</blockquote>
It sounds as thgough the DTMF tones are not being sent in a way that
Asterisk is seeing .....<br>
<br>
What type of telephone technology are you using? Hardware SIP
phones, software SIP phones, analogue phones via an FXS card,
analogue phones via a SIP ATA? What codec are you using?<br>
<br>
If you make an extension-to-extension call, can you send DTMF tones
down the line? Both ways around? Do they decode properly? (You
can get a mobile phone app for this.)<br>
<br>
<div class="gmail_extra">
<div class="gmail_quote"><span class="HOEnZb"><font
color="#888888">
<font color="#000000">--</font></font></span><br>
<span class="HOEnZb">
AJS</span><br>
<span class="HOEnZb">
</span><br>
<span class="HOEnZb">
Answers come *after* questions.</span><br>
<span class="HOEnZb">
</span>
<div class="HOEnZb">
<div class="h5"><br>
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