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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'>This may not apply to your situation, but
many ATAs and SIP phones have this feature built in to the device.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>We use Linksys/Sipura and auto redial and
last call return work without any special setup.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></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>Sherwood McGowan<br>
<b><span style='font-weight:bold'>Sent:</span></b> Monday, September 26, 2005
7:45 AM<br>
<b><span style='font-weight:bold'>To:</span></b> '<st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName>'<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Asterisk-Users] Call
Back On Busy?</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I know it's been touched on before, but no answers have been
found to the best of my knowledge. I'm using a SIP only setup, with a sip
provider giving PSTN and would like to see if anyone has an idea for creating
redial busy using ${DIALSTATUS} and possibly MeetMe?</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I figure something like this, but want to get feedback</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>1. Get callers last dialed number, if international number,
do not allow.</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>2. Playback a stuttertone to caller</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>3. Disconnect caller</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>4. Ring intended party check dial status. If busy,
wait 120 seconds and try again (do this for a total of 15 minutes)</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>5. If it's picked up, playback an announcement to the party
and put them in a meetme conference</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>6. Ring the original caller and bridge them to the meetme
conference. </span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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