<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40">
<head>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii">
<meta name=Generator content="Microsoft Word 12 (filtered medium)">
<style>
<!--
/* Font Definitions */
@font-face
        {font-family:"Cambria Math";
        panose-1:2 4 5 3 5 4 6 3 2 4;}
@font-face
        {font-family:Calibri;
        panose-1:2 15 5 2 2 2 4 3 2 4;}
@font-face
        {font-family:Tahoma;
        panose-1:2 11 6 4 3 5 4 4 2 4;}
@font-face
        {font-family:Verdana;
        panose-1:2 11 6 4 3 5 4 4 2 4;}
@font-face
        {font-family:"Segoe UI";
        panose-1:2 11 5 2 4 2 4 2 2 3;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
        {margin:0in;
        margin-bottom:.0001pt;
        font-size:12.0pt;
        font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
        {mso-style-priority:99;
        color:blue;
        text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
        {mso-style-priority:99;
        color:purple;
        text-decoration:underline;}
p
        {mso-style-priority:99;
        mso-margin-top-alt:auto;
        margin-right:0in;
        mso-margin-bottom-alt:auto;
        margin-left:0in;
        font-size:12.0pt;
        font-family:"Times New Roman","serif";}
p.MsoAcetate, li.MsoAcetate, div.MsoAcetate
        {mso-style-priority:99;
        mso-style-link:"Balloon Text Char";
        margin:0in;
        margin-bottom:.0001pt;
        font-size:8.0pt;
        font-family:"Tahoma","sans-serif";}
span.BalloonTextChar
        {mso-style-name:"Balloon Text Char";
        mso-style-priority:99;
        mso-style-link:"Balloon Text";
        font-family:"Tahoma","sans-serif";}
span.gmailquote
        {mso-style-name:gmail_quote;}
span.e
        {mso-style-name:e;}
span.EmailStyle22
        {mso-style-type:personal;
        font-family:"Verdana","sans-serif";
        color:windowtext;
        font-weight:normal;
        font-style:normal;
        text-decoration:none none;}
span.EmailStyle23
        {mso-style-type:personal-reply;
        font-family:"Verdana","sans-serif";
        color:windowtext;
        font-weight:normal;
        font-style:normal;
        text-decoration:none none;}
.MsoChpDefault
        {mso-style-type:export-only;
        font-size:10.0pt;}
@page Section1
        {size:8.5in 11.0in;
        margin:1.0in 1.0in 1.0in 1.0in;}
div.Section1
        {page:Section1;}
-->
</style>
<!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]-->
</head>
<body lang=EN-US link=blue vlink=purple>
<div class=Section1>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Some
additional information:<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>It
looks like it’s not even waiting for any additional DTMF signals as it
immediately tries to find the extension as soon as it gets the first digit:<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Sep
3 22:54:42 VERBOSE[15732] logger.c: -- Playing 'pbx-transfer' (language
'en')<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Sep
3 22:54:44 DEBUG[15738] chan_sip.c: * Detected inband DTMF '*'<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Sep
3 22:54:48 VERBOSE[15732] logger.c: -- Unable to find extension '*' in
context 'from-internal'<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Sep
3 22:54:48 VERBOSE[15732] logger.c: -- Playing 'pbx-invalid' (language
'en')<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Sep
3 22:54:52 VERBOSE[15732] logger.c: -- Stopped music on hold on
SIP/801-b190<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>I
was watching this log as it was generated … I dialed *801 but it only got the
first DTMF.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>This
was tested by picking up my SIP handset, dialing 7777, then dialing 801 which
did an outbound call to my cell phone. When I answered my cell phone, I pressed
# and was issued the prompt “Transfer” … I dialed *801, but as I said above it’s
not getting the additional DTMF tones.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>If
I dial *<b>in</b>* to the system from my cell phone and dial 801 it will send
me to the SIP extension. From there, I can successfully hit # and transfer the
call to *801 with no problems. It only seems to be when the call is
transferred to an external phone.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>This
isn’t an issue with the transfer system not finding the extension due to
improper context or something… this is an issue with Asterisk recognizing DTMF
tones during transfer.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>Any
thoughts?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>George<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #91C0FF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:9.0pt;font-family:"Segoe UI","sans-serif"'>From:</span></b><span
style='font-size:9.0pt;font-family:"Segoe UI","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>George A.
Roberts IV<br>
<b>Sent:</b> Sunday, September 03, 2006 10:06 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> RE: [asterisk-users] Help with blind transfer<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>This
isn’t A@H functionality. It’s basic Asterisk functionality.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'>George<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","sans-serif"'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #91C0FF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:9.0pt;font-family:"Segoe UI","sans-serif"'>From:</span></b><span
style='font-size:9.0pt;font-family:"Segoe UI","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>William
Piper<br>
<b>Sent:</b> Sunday, September 03, 2006 9:31 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] Help with blind transfer<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal>This is not the list for <a href="mailto:A@H">A@H</a>. For
questions about <a href="mailto:A@H">A@H</a> functionality, they have their own
mailing list.<br>
<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>bp<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><span class=gmailquote>On 9/3/06, <b>George A. Roberts IV</b>
<<a href="mailto:groberts@interjuncture.com">groberts@interjuncture.com</a>>
wrote:</span> <o:p></o:p></p>
<div>
<div>
<div>
<p>No one has any ideas?<o:p></o:p></p>
<p> <o:p></o:p></p>
<div>
<div style='border:none;border-top:solid #91C0FF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p><b><span style='font-size:9.0pt'>From:</span></b><span style='font-size:
9.0pt'> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com
</a>[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com"
target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>George
A. Roberts IV <br>
<b>Sent:</b> Friday, September 01, 2006 10:37 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> [asterisk-users] Help with blind transfer</span><o:p></o:p></p>
</div>
</div>
</div>
<div>
<p> <o:p></o:p></p>
<p>Hello all,<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>Using Asterisk@Home and are using its Follow Me functionality to bounce
calls to my cell phone when I'm not in the office. Would like to be able
to use the blind transfer functionality from my cell phone when I receive a
call in from Asterisk but am not having much luck getting it to work… <o:p></o:p></p>
<p> <o:p></o:p></p>
<p>I can press ## (that's what it's set to in features.conf) and get the
"Transfer" prompt from Alison and the dialtone. But no matter
what I punch in, it seems that Asterisk is only getting the first digit I
press.<o:p></o:p></p>
<p> <o:p></o:p></p>
<p> -- Unable to find extension '*' in context
'from-internal'<o:p></o:p></p>
<p> -- Playing 'pbx-invalid' (language 'en')<o:p></o:p></p>
<p> -- Stopped music on hold on SIP/801-b190<o:p></o:p></p>
<p> -- Started music on hold, class 'default', on
SIP/801-b190<o:p></o:p></p>
<p> -- Playing 'pbx-transfer' (language 'en')<o:p></o:p></p>
<p> -- Unable to find extension '8' in context
'from-internal'<o:p></o:p></p>
<p> -- Playing 'pbx-invalid' (language 'en')<o:p></o:p></p>
<p> <o:p></o:p></p>
<p> <o:p></o:p></p>
<p>The first one there I tried to punch in *801 to transfer to voicemail.
The second one I punched in 802 to transfer to another extension. Each
time, it's only getting the first digit.<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>Anyone seen this before? Any thoughts?<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>Thanks!<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>Regards,<o:p></o:p></p>
<p> <o:p></o:p></p>
<p>George A. Roberts IV<o:p></o:p></p>
<p>President & CEO, Interjuncture Corp.<o:p></o:p></p>
<p><a href="http://www.interjuncture.com/" target="_blank">http://www.interjuncture.com/</a><o:p></o:p></p>
<p> <o:p></o:p></p>
</div>
</div>
</div>
<p class=MsoNormal><br>
_______________________________________________<br>
--Bandwidth and Colocation provided by <a href="http://easynews.com/"
target="_blank">Easynews.com </a>--<br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users"
target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<br>
<br>
<br clear=all>
<o:p></o:p></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
</body>
</html>