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<DIV dir=ltr align=left><SPAN class=576114002-31012006><FONT face=Arial
color=#0000ff size=2>It can be done.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=576114002-31012006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=576114002-31012006><FONT face=Arial
color=#0000ff size=2>1. Setup a new Vm profile on CCM with a mask of XXXX
</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=576114002-31012006><FONT face=Arial
color=#0000ff size=2>2. Setup a CTI route point:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=576114002-31012006> <FONT
face=Arial color=#0000ff size=2>a. Set the directory number to a
pattern. I use *27XX</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=576114002-31012006> <FONT
face=Arial color=#0000ff size=2>but any pattern that you can send from * is
good, ie. 88XXX</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=576114002-31012006> <FONT
face=Arial color=#0000ff size=2>b. Set the VM profile to the newly created
profile</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=576114002-31012006></SPAN><SPAN
class=576114002-31012006><FONT face=Arial color=#0000ff
size=2> c. Set the line to forward all calls to
VM</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=576114002-31012006><FONT face=Arial
color=#0000ff size=2>3. Change the dialplan in * to append the extension
called to</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=576114002-31012006> <FONT
face=Arial color=#0000ff size=2>the prefix pattern.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=576114002-31012006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=576114002-31012006><FONT face=Arial
color=#0000ff size=2><FONT face="Times New Roman" color=#000000 size=3>exten
=> 123,1,Dial(SIP/sipphone,20)<BR>exten =>
123,2,Dial(SIP/ccm/88123)</FONT></FONT></SPAN></DIV>
<DIV><SPAN class=576114002-31012006><FONT face=Arial color=#0000ff size=2><FONT
face=Arial color=#0000ff size=2></FONT></FONT></SPAN> </DIV>
<DIV><SPAN class=576114002-31012006><FONT face=Arial color=#0000ff size=2><FONT
face="Times New Roman" color=#000000 size=3>
<DIV dir=ltr align=left><SPAN class=576114002-31012006><FONT face=Arial
color=#0000ff size=2>I use this setup to allow my CCM users to transfer calls
directly to VM,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=576114002-31012006></SPAN></FONT><SPAN
class=576114002-31012006>but it should work for this purpose as
well.</SPAN></DIV><SPAN class=576114002-31012006></SPAN></FONT></SPAN></DIV>
<DIV><SPAN class=576114002-31012006><SPAN class=576114002-31012006></SPAN>
<DIV dir=ltr align=left><SPAN class=576114002-31012006></SPAN><FONT face=Arial
color=#0000ff size=2>D<SPAN
class=576114002-31012006>an</SPAN><BR></FONT></DIV></DIV></SPAN><BR>
<BLOCKQUOTE style="MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<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>sys
read<BR><B>Sent:</B> Tuesday, January 24, 2006 7:57 AM<BR><B>To:</B> Asterisk
Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> Re:
[Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM4.0SIP
Trunk<BR></FONT><BR></DIV>
<DIV></DIV><BR>I have my eyes on the Linksys/Sipura 941, ( SIP ), but the core
problem is that you can't use SIP phones with CCM. I have a SIP trunk
between asterisk and ccm. I can route calls back and forth, I just can't
get the call to send to vm if no answer on the asterisk side. <BR><BR><BR>
<DIV><SPAN class=gmail_quote>On 1/24/06, <B class=gmail_sendername>kevin
ling</B> <<A
href="mailto:kevin.ling@amphenol.com.tw">kevin.ling@amphenol.com.tw</A>>
wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">
<DIV dir=ltr align=left><SPAN><FONT face=Arial color=#0000ff
size=2>Hi,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN><FONT face=Arial color=#0000ff size=2>Maybe
buy 7912 phone and register to CCM is another choice. or integrated CCM with
asterisk voicemail system.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN></SPAN><SPAN></SPAN><SPAN><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left>
<HR>
<FONT face=Tahoma size=2><SPAN class=q><B>From:</B> <A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:asterisk-users-bounces@lists.digium.com"
target=_blank>asterisk-users-bounces@lists.digium.com </A>[mailto:<A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:asterisk-users-bounces@lists.digium.com"
target=_blank>asterisk-users-bounces@lists.digium.com</A>] <B>On Behalf Of
</B>sys read<BR></SPAN><B>Sent:</B> Tuesday, January 24, 2006 11:28
PM<BR><B>To:</B> Asterisk Users Mailing List - Non-Commercial
Discussion<BR><B>Subject:</B> Re: [Asterisk-Users] Asterisk SIP phones to
Cisco Unity via CCM 4.0SIP Trunk<BR></FONT><BR></DIV>
<DIV><SPAN class=e id=q_108fd1d992db90cd_3>
<DIV></DIV><BR>Hi guys,<BR><BR>I want to leave messages on our unity
box. I have already converted a couple 7940s to SIP, but I can't
give them out to our users because I don't want to have to deal with two
voicemail systems.<BR><BR>we have licenses for all our users on unity as
is. we're about to buy a bunch more 7940s, but I don't
want to cause they're expensive. I'd rather buy a
cheaper SIP phone and have it rollover to the unity vm.<BR><BR>
<DIV><SPAN class=gmail_quote>On 1/23/06, <B class=gmail_sendername>Gary
Richardson</B> <<A onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:gary.richardson@gmail.com"
target=_blank>gary.richardson@gmail.com</A>> wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">You
can run a SIP image on a 7940. Asterisk@home has pretty good<BR>support
for it. Check the <A
onclick="return top.js.OpenExtLink(window,event,this)"
href="http://voip-info.org" target=_blank>voip-info.org</A> wiki for
instructions on<BR>switching the firmware.<BR><BR>Hopefully that will take
a step out of the plan -- you could <BR>completely ditch your Cisco system
:)<BR><BR>On 1/23/06, sys read <<A
onclick="return top.js.OpenExtLink(window,event,this)"
href="mailto:sysread@gmail.com" target=_blank>sysread@gmail.com</A>>
wrote:<BR>><BR>> Hi,<BR>><BR>> I've got a CCM ( Cisco Call
Manager ), with a Cisco Unity VM server and <BR>> about 45 SCCP phones
on the ccm, and 200 users on unity. we want to<BR>> migrate
all users to IP Phones to ditch our ancient phone system. I
would<BR>> love to get Linksys-Sipura SPA-941s for the 150 users not on
IP phones yet <BR>> and run sip to an asterisk server, but have their
voicemail on Unity.<BR>><BR>> these phones are $150 each, the
alternative is cisco 7940s ( around $250 )<BR>> running SCCP through
the CCM. at the quantities I'm talking about, $100 is <BR>>
significant.<BR>><BR>> Does anyone have any idea how to get this
done?<BR>><BR>> I've tried this:<BR>><BR>> exten =>
123,1,Dial(SIP/sipphone,20)<BR>> exten =>
123,2,Dial(SIP/ccm/3040)<BR>><BR>> where 3040 is our VM pilot for
ccm. but all it does is take us to the main<BR>>
greeting.<BR>><BR>> we have smartnet, but they haven't been helpful
at all<BR>><BR>> I called digium to see if they could help if we
paid, but they said they've <BR>> never heard of cisco
unity....<BR>><BR>> help?<BR>><BR>> thanks.<BR>><BR>>
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