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<DIV><FONT face=Arial size=2>Thank you, For the responces i had dtmfmode=inband
when rcf2833 was the proper setting. I feel retarded that i missed that, but it
happens. thanks again</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Tim Touhsaent</FONT></DIV>
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<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=igil@europesip.com
href="mailto:igil@europesip.com">igil@europesip.com</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">Asterisk Users Mailing List -
Non-Commercial Discussion</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Friday, April 29, 2005 9:38
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Asterisk-Users] need
help</DIV>
<DIV><BR></DIV><BR><FONT face=sans-serif size=2>This is a DTMF issue,</FONT>
<BR><BR><FONT face=sans-serif size=2>You must adjust this on the especific
channel conf file.</FONT> <BR><BR><FONT face=sans-serif size=2>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> <BR><BR><FONT
face=sans-serif size=2>Ismael.</FONT> <BR><BR><BR><BR><BR><BR>
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<TD width="40%"><FONT face=sans-serif size=1><B>"Tim Touhsaent" <<A
href="mailto:touhsatj@hotmail.com">touhsatj@hotmail.com</A>></B>
</FONT><BR><FONT face=sans-serif size=1>Enviado por: <A
href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</A></FONT>
<P><FONT face=sans-serif size=1>04/29/2005 03:16 PM</FONT>
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<TD bgColor=white><FONT face=sans-serif size=1>Por favor, responda
a<BR>Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com></FONT></TR></TBODY></TABLE><BR></P>
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<TD><FONT face=sans-serif size=1>Para</FONT> <BR>
<TD><FONT face=sans-serif size=1>"Asterisk Users Mailing List -
Non-Commercial Discussion"
<asterisk-users@lists.digium.com></FONT>
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<TD><FONT face=sans-serif size=1>cc</FONT>
<TD>
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<TD>
<TD>
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<TD><FONT face=sans-serif size=1>Asunto</FONT>
<TD><FONT face=sans-serif size=1>[Asterisk-Users] need
help</FONT></TR></TBODY></TABLE><BR>
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<TD></TR></TBODY></TABLE><BR></TR></TBODY></TABLE><BR><BR><BR><FONT
face="Courier New" size=2>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><BR>Tim<BR>_______________________________________________<BR>Asterisk-Users
mailing
list<BR>Asterisk-Users@lists.digium.com<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>To
UNSUBSCRIBE or update options visit:<BR>
http://lists.digium.com/mailman/listinfo/asterisk-users<BR></FONT><BR><BR>
<P>
<HR>
<P></P>_______________________________________________<BR>Asterisk-Users
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list<BR>Asterisk-Users@lists.digium.com<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>To
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