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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'>In my patching, I have only dealt with
(enabled) a single (first) line appearance.<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><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>------------------------------- <br>
<strong><b><font face=Arial><span style='font-family:Arial'>Eugene A. Grossi,
M.D.</span></font></b></strong><br>
Professor of Cardiothoracic Surgery<br>
<st1:place w:st="on"><st1:PlaceName w:st="on">New York</st1:PlaceName> <st1:PlaceType
w:st="on">University</st1:PlaceType> <st1:PlaceType w:st="on">School</st1:PlaceType></st1:place>
of Medicine<br>
Suite 9-V; <st1:Street w:st="on"><st1:address w:st="on">530 First Avenue</st1:address></st1:Street><br>
<st1:place w:st="on"><st1:City w:st="on">New York</st1:City>, <st1:State w:st="on">NY</st1:State>
<st1:PostalCode w:st="on">10016</st1:PostalCode></st1:place><br>
ph (212) 263-7452<br>
fax(212) 263-5534<br>
<st1:PersonName w:st="on">grossi@cv.med.nyu.edu</st1:PersonName></span></font><o:p></o:p></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'> Raj Jain
[mailto:rj2807@gmail.com] <br>
<b><span style='font-weight:bold'>Sent:</span></b> Friday, January 04, 2008
5:07 AM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Developers Mailing
List<br>
<b><span style='font-weight:bold'>Cc:</span></b> <st1:PersonName w:st="on">grossi@cv.med.nyu.edu</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-dev]
Implementation of Broadsoft Sip Access in Asterisk to enable <st1:place w:st="on">SLA</st1:place>
for Sipura/Linksys</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>
<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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<blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;
margin-left:4.8pt;margin-right:0in'>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Somewhere in my mail archive I have a permission from Broadsoft or <br>
Cylantro to implement this - before they<br>
published it as a draft. They wanted Asterisk to include this<br>
functionality. After that, they published it openly<br>
as a draft, so there should be no problems.<o:p></o:p></span></font></p>
</blockquote>
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<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'><br>
It's probably also supported by the fact that this specification is publicly
available on BroadSoft's company website: <br>
<a
href="http://www.broadsoft.com/pdf/BW-SIPAccessSideExtensionsInterfaceSpec-R1301.pdf">http://www.broadsoft.com/pdf/BW-SIPAccessSideExtensionsInterfaceSpec-R1301.pdf</a><br>
<br>
If we speak about <st1:place w:st="on">SLA</st1:place> specifically, they've
basically extended SIP on three fronts:<br>
<br>
1. Additional parameters in the existing Call-Info header <br>
2. Call-Info SIP event-package<br>
3. Line-Seize SIP event-package<br>
<br>
The Call-Info event-package pretty much accomplishes the same thing as the
standard dialog event-package (i.e. it notifies the state changes of the shared
line to <st1:place w:st="on">SLA</st1:place> stations). <br>
<br>
The Line-Seize event-package solves an arbitration issue that happens in
en-bloc signaling protocols such as SIP. It allows the entire SLA group to be
immediately informed as soon as one <st1:place w:st="on">SLA</st1:place>
station seizes the shared line. This allows an <st1:place w:st="on">SLA</st1:place>
station to "win" the line. The other phones render the line busy
while one phone is dialing. I've seen different commercial PBX vendors solve
this problem in their own proprietary way. <br>
<br>
There is one more interesting aspect to these SIP extensions -- they're built
on the concept of Call Appearance. Call Appearance means that a unique line can
have multiple calls active on it at the same time. So, you're not only telling
the phones which line underwent a state change, but also which appearance of
it. <br>
<br>
Does the current Asterisk SLA implementation support the Call Appearance
concept?<br>
<br>
--<br>
Raj<o:p></o:p></span></font></p>
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