<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="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 11 (filtered medium)">
<!--[if !mso]>
<style>
v\:* {behavior:url(#default#VML);}
o\:* {behavior:url(#default#VML);}
w\:* {behavior:url(#default#VML);}
.shape {behavior:url(#default#VML);}
</style>
<![endif]-->
<title>Message</title>
<style>
<!--
/* Font Definitions */
@font-face
{font-family:Wingdings;
panose-1:5 0 0 0 0 0 0 0 0 0;}
@font-face
{font-family:Tahoma;
panose-1:2 11 6 4 3 5 4 4 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman";}
a:link, span.MsoHyperlink
{color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{color:purple;
text-decoration:underline;}
span.EmailStyle17
{mso-style-type:personal-reply;
font-family:Arial;
color:navy;}
@page Section1
{size:8.5in 11.0in;
margin:1.0in 1.25in 1.0in 1.25in;}
div.Section1
{page:Section1;}
-->
</style>
</head>
<body lang=EN-US link=blue vlink=purple>
<div class=Section1>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I don’t think the nature of these
phones would allow for such a thing. It was designed for transfers and such, to
be a real PBX, not like having 4 phone lines from BellSouth and multiple 4 line
phones.<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'>I couldn’t imagine SER/Asterisk/any
SIP proxy or program doing what is needed.<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'>The only idea I had to get asterisk to do it
would be have the calling party thrown into a conference room right away, and
then have it ring all the other phones. Whoever answers it would then be put
into the conference room with the calling party. But I think the trick is,
whenever a person calls in, they get thrown put into a conference room, and
then the PolyCom’s all have to auto-answer and place the calls on silent
hold, so that everybody is thrown into the conference room. That shouldn’t
be TOO hard to rig, but how do you get all the phones to ring as well until
somebody picks up, so that there is at least 1 active person in the conference
with the calling party. Then any other phone should be able to bust in
simply by taking that line off of hold. <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'>Good luck with that </span></font><font
size=2 color=navy face=Wingdings><span style='font-size:10.0pt;font-family:
Wingdings;color:navy'>J</span></font><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'> <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>
<div>
<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>
<hr size=2 width="100%" align=center tabindex=-1>
</span></font></div>
<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>David Gomillion<br>
<b><span style='font-weight:bold'>Sent:</span></b> Friday, November 19, 2004
12:19 PM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Asterisk-Users] Shared
line appearances</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>All right. It would appear that I am not the only one
interested in shared line appearances. Many others have stated that they
wish for the key-system-like feature of the blinking lights. Quite
frankly, I don't think it's a good thing, but the people who use these systems
are very resistant to change. I set up call queues, pickup groups,
1-touch transfers, and still nothing seems to placate them. If I could, I
would just replace the users...</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I mainly use Polycom SoundPoint IP phones: some 300s and
some 600s. Bottom line is that if I am going to be able to finish the
rollout of the phone system, and switch away from having 2 PBXs vying for power
(Asterisk and the Nortel NorStar MICS system), I am going to have to get this
feature working. I have received authorization to offer a bounty to get
it working in Asterisk, and to then contribute the source to the project.</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>As I have studied the issue, I'm not sure it is within the
"master plan" for asterisk. Searching the archives, it seems we
only expect Asterisk to be a "clever UA". The people asking
were advised to get a real SIP proxy. In passing, someone asked if
chan_sip2 would support it, but I found no response.</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Many references to SER have been made. I have
installed SER successfully. I then tried to make the feature work, but
have been unsuccessful. Both lines will ring, but the first person to
answer the call gets it, and the other phone's lights are as dark as can
be. SER does not seem to do any better with "line-seize" than
Asterisk. At least Asterisk has the hint to allow the lights to work (I
have not yet implemented this, but since it does not meet the requirements, it
does not really matter)... but neither system will allow the caller to press
the blinking light on a call that was placed on hold to answer it.</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I am now looking at other SIP proxies. I am in the
process of installing sipXpbx, which includes many different pieces of the
Pingtel sipExpress system that have been open-sourced. I am not sure
which pieces I will need specifically, so I will install the whole shooting
match and see if the feature even works. If it does, I'll remove packages
and try to reintegrate with Asterisk.</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Has anyone gotten shared line appearances to work with
Polycom Soundpoint IP phones? Not just blinking lights, but the whole
shebang: lights, pressing the button to seize the line, shared registrations,
etc. Is it best to work with a 3rd party SIP proxy/router/whatever, or
should we pool resources and get the feature integrated into Asterisk somehow?</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Looking forward to your thoughts,</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>David Gomillion</span></font><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
</div>
</div>
</body>
</html>