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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Hi All,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>My company has a customer who wishes to integrate an IVR
system into another system, and Asterisk seems to meet the needs very well from
the IVR perspective.<span style='mso-spacerun:yes'> </span>As part of the
integration effort, think we will need to do some custom modifications to
Asterisk, and will also need help in configuring the Asterisk installation to
do the basic IVR stuff.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>The high-level scenario is as follows:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<ol style='mso-margin-top-alt:0cm' start=1 type=1>
<li class=MsoNormal style='mso-list:l1 level1 lfo5;tab-stops:list 36.0pt'><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>Our
customer operates a <span class=SpellE>VoIP</span> service using a soft
switch that does not have IVR functionality - this is what we need
to integrate the Asterisk system to.<o:p></o:p></span></font></li>
<li class=MsoNormal style='mso-list:l1 level1 lfo5;tab-stops:list 36.0pt'><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>The
existing <span class=SpellE>VoIP</span> system initiates a call by
accepting an SMS with a destination number (party B) from a registered
user (party A).<span style='mso-spacerun:yes'> </span>The <span
class=SpellE>VoIP</span> system then initiates *<b><span style='font-weight:
bold'>two</span></b>* calls - one from the <span class=SpellE>VoIP</span>
system to registered user A, and another from the <span class=SpellE>VoIP</span>
system to the destination number B, and then binds the two calls together.<o:p></o:p></span></font></li>
<li class=MsoNormal style='mso-list:l1 level1 lfo5;tab-stops:list 36.0pt'><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>If the
destination number B does not answer, the customer wants the <span
class=SpellE>VoIP</span> system to connect the registered user A to an IVR
system to record a message for B to recover via a web site.<o:p></o:p></span></font></li>
<li class=MsoNormal style='mso-list:l1 level1 lfo5;tab-stops:list 36.0pt'><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>The
problem is that Asterisk will not know anything about party B who did not
pick up when the <span class=SpellE>VoIP</span> system made the call.<span
style='mso-spacerun:yes'> </span>Also, from the point of view of
Asterisk, calls will always be coming from the <span class=SpellE>VoIP</span>
system, and therefore the same "number".<span
style='mso-spacerun:yes'> </span>Once a message has been recorded, how
do we identify the recorded voice messages in order to deliver them via a
separate web page at a later date?<o:p></o:p></span></font></li>
<li class=MsoNormal style='mso-list:l1 level1 lfo5;tab-stops:list 36.0pt'><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>There
are several hundred thousand registered members using the <span
class=SpellE>VoIP</span> system currently, and there are a lot of new
users coming onboard all the time.<o:p></o:p></span></font></li>
</ol>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Our proposed solution is:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<ol style='mso-margin-top-alt:0cm' start=1 type=1>
<li class=MsoNormal style='mso-list:l2 level1 lfo6;tab-stops:list 36.0pt'><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>When
connecting the user A with the Asterisk IVR system if party B does not
answer, the <span class=SpellE>VoIP</span> system will pass the phone
number of B as the "caller" number when calling the IVR (in
other words the <span class=SpellE>VoIP</span> system will fake the sender
number)<o:p></o:p></span></font></li>
<li class=MsoNormal style='mso-list:l2 level1 lfo6;tab-stops:list 36.0pt'><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>Asterisk
(custom coding here?) will use the caller number as part of the naming convention
of the recorded voice message so it can be retrieved later<o:p></o:p></span></font></li>
<li class=MsoNormal style='mso-list:l2 level1 lfo6;tab-stops:list 36.0pt'><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>A web
site (that we will have built) will retrieve the message file using the party
B number and make it available for download by the user B<o:p></o:p></span></font></li>
<li class=MsoNormal style='mso-list:l2 level1 lfo6;tab-stops:list 36.0pt'><font
size=2 face=Arial><span style='font-size:10.0pt;font-family:Arial'>This
service will be available "member to member" so we will know
who the caller is, and how to let them know they have a message waiting.<o:p></o:p></span></font></li>
</ol>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>We are an Australian organisation, so we would prefer to
contract resources here, but are open to engaging internationally. We have moderate
Linux skills and high level J2EE skills in-house, but very little experience in
Perl, and insufficient time to get up to speed to develop the needed changes to
Asterisk. If the needed changes are appropriate to contribute to the Open
Source community, then we are happy to do so. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I would appreciate expressions of interest from people who
would be prepared to do the work configuring Asterisk as an IVR, and doing
whatever development work is needed to retrieve the messages.<span
style='mso-spacerun:yes'> </span>I realise that this is a very high level
outline of the job, and indeed there may well be a better way of doing it, so
alternate suggestions are welcome.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Cheers,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Adam Wells<o:p></o:p></span></font></p>
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