<br><font size=2 face="sans-serif">This is a DTMF issue,</font>
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<br><font size=2 face="sans-serif">You must adjust this on the especific channel conf file.</font>
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<br><font size=2 face="sans-serif">For example, ia sip phone cannot dial any number during an active call, you must see sip.conf and the config in your hardphone or softphone.</font>
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<br><font size=2 face="sans-serif">Ismael.</font>
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<td width=40%><font size=1 face="sans-serif"><b>"Tim Touhsaent" <touhsatj@hotmail.com></b> </font>
<br><font size=1 face="sans-serif">Enviado por: asterisk-users-bounces@lists.digium.com</font>
<p><font size=1 face="sans-serif">04/29/2005 03:16 PM</font>
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<td bgcolor=white><font size=1 face="sans-serif">Por favor, responda a<br>
Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com></font></table>
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<td><font size=1 face="sans-serif">Para</font>
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<td><font size=1 face="sans-serif">"Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com></font>
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<td><font size=1 face="sans-serif">Asunto</font>
<td><font size=1 face="sans-serif">[Asterisk-Users] need help</font></table>
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<br><font size=2 face="Courier New">I am having an issue with the asterisk system not responding to dialed<br>
numbers during an active<br>
call. I'm not even sure where to look, zapata.conf? sip.conf? or the phone<br>
config? and worse I<br>
don't even know what Keywords to search for.<br>
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Tim<br>
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