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<DIV dir=ltr align=left><SPAN class=781152914-26092005><FONT face=Arial
color=#0000ff size=2>Thank you, I do appreciate that many ATAs have redial on
busy, but I've been given the charge of figuring out how one would do it in
Asterisk.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=781152914-26092005><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=781152914-26092005><FONT face=Arial
color=#0000ff size=2>Don't ask me why</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=781152914-26092005></SPAN> </DIV><BR>
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style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
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<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Damon
Estep<BR><B>Sent:</B> Monday, September 26, 2005 10:15 AM<BR><B>To:</B>
Asterisk Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> RE:
[Asterisk-Users] Call Back On Busy?<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=Section1>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">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 face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">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 face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
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<P class=MsoNormal><B><FONT face=Tahoma size=2><SPAN
style="FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">From:</SPAN></FONT></B><FONT
face=Tahoma size=2><SPAN style="FONT-SIZE: 10pt; 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></DIV>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"><o:p> </o:p></SPAN></FONT></P>
<DIV>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; 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></DIV>
<DIV>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"> <o:p></o:p></SPAN></FONT></P></DIV>
<DIV>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">I figure something like this, but
want to get feedback</SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"> <o:p></o:p></SPAN></FONT></P></DIV>
<DIV>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">1. Get callers last dialed number,
if international number, do not allow.</SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">2. Playback a stuttertone to
caller</SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">3. Disconnect
caller</SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; 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></DIV>
<DIV>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; 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></DIV>
<DIV>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">6. Ring the original caller and
bridge them to the meetme conference. </SPAN></FONT><o:p></o:p></P></DIV>
<DIV>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"> <o:p></o:p></SPAN></FONT></P></DIV></DIV></DIV></BLOCKQUOTE></BODY></HTML>