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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Your dial string must have either the t or
T option set.</span></font></p>
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10.0pt;font-family:Arial;color:navy'> </span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>omar parihuana<br>
<b><span style='font-weight:bold'>Sent:</span></b> Friday, December 01, 2006
9:10 AM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion<br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users] CALL
TRANSFER</span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Hi Guys,</span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>I'm implementing my Asterisk step by step, so far the communications
between softphones, hardphones with Gateways, voice mail, are working fine.
Rightnow I would like to enable Call tranfer (like Traditional PBX) in Blind
Transfer and AttendXFER, I'm reading features.conf in accordance to <a
href="http://voip-info.org">voip-info.org</a> but the transfer doesn't
work! Please if you can provide me some examples will be very appreciate.</span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Rgds.<br clear=all>
<br>
-- <br>
Omar E.P.T<br>
-----------------<br>
Certified Networking Professionals make better Connections!<br>
<br>
<a href="http://omarept.blogspot.com/">http://omarept.blogspot.com/</a><br>
<br>
Usysnet Corp <br>
Open Source Solutions<br>
<a href="http://www.usysnet.com.pe">www.usysnet.com.pe</a> </span></font></p>
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