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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I experiment the same problem… On some call, there’s
no possibility to transfer the call (caller ear DTMF sounds) – AST 1.4.21.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>For Max : <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>DialPlan must be like this :<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal style='text-indent:35.4pt'><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'>exten => 99, 1, Answer()<o:p></o:p></span></p>
<p class=MsoNormal style='text-indent:35.4pt'><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'>exten => 99, n, Dial(SIP/sip_user1,
20, tT)<o:p></o:p></span></p>
<p class=MsoNormal style='text-indent:35.4pt'><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'>exten => 99, 1, Hangup()<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Regards<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span lang=FR style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>De :</span></b><span
lang=FR style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-dev-bounces@lists.digium.com [mailto:asterisk-dev-bounces@lists.digium.com]
<b>De la part de</b> Fernando Urzedo<br>
<b>Envoyé :</b> lundi, 25. août 2008 13:49<br>
<b>À :</b> Asterisk Developers Mailing List<br>
<b>Objet :</b> Re: [asterisk-dev] Blind Transfer is not working in
incoming calls<o:p></o:p></span></p>
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<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Hi Max,</span><o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Please make sure you are setting the Dial command with parameters
"T" and "t". I would say you are missing "t"...</span><o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Regards!</span><o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
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<hr size=2 width="100%" align=center>
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<p class=MsoNormal style='margin-bottom:12.0pt'><b><span lang=EN-US
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-dev-bounces@lists.digium.com
[mailto:asterisk-dev-bounces@lists.digium.com] <b>On Behalf Of </b>Max Alex<br>
<b>Sent:</b> sábado, 23 de agosto de 2008 03:53<br>
<b>To:</b> asterisk-dev@lists.digium.com<br>
<b>Subject:</b> [asterisk-dev] Blind Transfer is not working in incoming calls</span><span
lang=EN-US><o:p></o:p></span></p>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'>Hi Everybody,<br>
i have installed asterisk 1.4.19 on my box,<br>
I have setup agi script which is used while incoming and outgoing calls.<br>
It will find the users for incoming and calls to them which is registered in
asterisk,<br>
I have a setup *# for blind transfer to call any outbound or inbound numbers.<br>
when i am calling any outbound call and the calls are connected with my sip
peer, then i am pressing *# for blind transfer, it will ask me to enter the
transfer number and it is working,<br>
But when an incoming call to my sip user and they are connected the *# is not
worked even the transfer prompt is also played, and dtmf is also set properly.<br>
<br>
But i am not getting why the incoming call is not transfer to any other number?<br>
Please help for this issue!<br clear=all>
<br>
<br>
-- <br>
Thanks,<br>
Max Alex<br>
Voip Developer<o:p></o:p></p>
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