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<DIV dir=ltr align=left><FONT face=Arial color=#0000ff size=2><SPAN
class=773494219-30012009>Does a normal Unity server have "voice cards"?
I'm starting an installation and I think this is the first time I've seen voice
cards mentioned.</SPAN></FONT></DIV>
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<P align=left><STRONG>Steve Miller</STRONG><BR>Telecom Engineer<BR>Dickstein
Shapiro LLP<BR>1825 Eye Street NW | Washington, DC 20006<BR>Tel (202) 420-3370|
Fax (202) 330-5607<BR><A
href="mailto:MillerS@dicksteinshapiro.com">MillerS@dicksteinshapiro.com</A><FONT
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<FONT face=Tahoma size=2><B>From:</B> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <B>On Behalf Of </B>Lisa
Notarianni<BR><B>Sent:</B> Friday, January 30, 2009 1:35 PM<BR><B>Cc:</B>
cisco-voip@puck.nether.net<BR><B>Subject:</B> Re: [cisco-voip] Conversion from
Centrex to VoIP<BR></FONT><BR></DIV>
<DIV></DIV>Thanks all! That did the trick!<BR><BR>Lisa<BR><BR>Lelio
Fulgenzi wrote:
<BLOCKQUOTE
cite=mid:31267827.4499101233334942595.JavaMail.root@superior.cs.uoguelph.ca
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<DIV style="FONT-SIZE: 10pt; COLOR: rgb(0,0,0); FONT-FAMILY: Verdana">This is
the model we chose. You will need a route pattern which sends calls to the
PBX/Centrex accordingly.<BR><BR>If you are using 6 or higher, then you can
leave them in the partition and have forward on unregistered to their
DID.<BR><BR><BR>---<BR>Lelio Fulgenzi, B.A.<BR>Senior Analyst (CCS) *
University of Guelph * Guelph, Ontario N1G 2W1<BR>(519) 824-4120 x56354 (519)
767-1060 FAX
(JNHN)<BR>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<BR>"Bad
grammar makes me [sic]" - Tshirt<BR><BR><BR>----- Original Message
-----<BR>From: "Lisa Notarianni" <A class=moz-txt-link-rfc2396E
href="mailto:notariannil1@scranton.edu"><notariannil1@scranton.edu></A><BR>To:
"Carlos Ortiz" <A class=moz-txt-link-rfc2396E
href="mailto:COrtiz@sscincorporated.com"><COrtiz@sscincorporated.com></A><BR>Cc:
<A class=moz-txt-link-abbreviated
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</A><BR>Sent:
Friday, January 30, 2009 11:28:45 AM GMT -05:00 US/Canada Eastern<BR>Subject:
Re: [cisco-voip] Conversion from Centrex to VoIP<BR><BR>I had this similar
response from a few which I thought would be an option but won't VoIP users
get an error when trying to call them? They need to be able to use
Centrex until I am ready to make their VoIP phone live. <BR><BR>Carlos
Ortiz wrote:
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<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Add them to a
temporary partition that no one has access to via Calling Search
Spaces. Then when you want to activate them use BAT to add them to the
real partition they belong to. This will allow you to add them to Call
Mgr ahead of time but not be accessible from the already converted VoIP
phones….</SPAN></FONT></P>
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style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT> </P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Carlos</SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"></SPAN></FONT> </P>
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<P class=MsoNormal><B><FONT face=Tahoma color=black size=2><SPAN
style="FONT-WEIGHT: bold; FONT-SIZE: 10pt; COLOR: windowtext; FONT-FAMILY: Tahoma">From:</SPAN></FONT></B><FONT
face=Tahoma color=black size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: windowtext; FONT-FAMILY: Tahoma"> <A
class=moz-txt-link-abbreviated
href="mailto:cisco-voip-bounces@puck.nether.net" target=_blank
moz-do-not-send="true">cisco-voip-bounces@puck.nether.net</A> [<A
class=moz-txt-link-freetext href="mailto:cisco-voip-bounces@puck.nether.net"
target=_blank
moz-do-not-send="true">mailto:cisco-voip-bounces@puck.nether.net</A>]
<B><SPAN style="FONT-WEIGHT: bold">On Behalf Of </SPAN></B>Lisa
Notarianni<BR><B><SPAN style="FONT-WEIGHT: bold">Sent:</SPAN></B> Friday,
January 30, 2009 11:13 AM<BR><B><SPAN
style="FONT-WEIGHT: bold">Cc:</SPAN></B> <A class=moz-txt-link-abbreviated
href="mailto:cisco-voip@puck.nether.net" target=_blank
moz-do-not-send="true">cisco-voip@puck.nether.net</A><BR><B><SPAN
style="FONT-WEIGHT: bold">Subject:</SPAN></B> [cisco-voip] Conversion from
Centrex to VoIP</SPAN></FONT><FONT color=black><SPAN
style="COLOR: windowtext"></SPAN></FONT></P></DIV>
<P class=MsoNormal><FONT face="Times New Roman" color=black size=3><SPAN
style="FONT-SIZE: 12pt"></SPAN></FONT> </P>
<DIV>
<P class=MsoNormal><FONT face="Times New Roman" color=black size=3><SPAN
style="FONT-SIZE: 12pt">I asked this question before but did not get a
response from anyone. I am going to reword it...<BR><BR>We are in the
process of switching from Centrex to VoIP. As we move along and
convert each department, we find timing difficult because as soon as the
phones are entered in Call Manager, those already on VoIP calling the lines
on that phone get an error since the phone is not yet plugged in.<BR><BR>In
addition to having convert the lines from Centrex to our DID block (I can
work around that), we find it difficult to come up with a solution with
being able to work ahead of schedule by adding the phones into Call Manager
without having calling issues from VoIP phones. <BR><BR>Anyone know of
a creative way to add these phones to Call Manager but isolate them from
other VoIP phones until we are ready to deploy them to desks?
<BR><BR>Thanks,<BR><BR>Lisa<BR><BR><BR><IMG id=_x0000_i1025 height=100
src="cid:part1.06060308.05040007@scranton.edu" width=250 border=0
moz-do-not-send="true"></SPAN></FONT></P></DIV></DIV>
<DIV>
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