Greg,<br><br>appending the number just gives me a fast busy.<br><br>Mike,<br><br>a) is out because the cheaper cisco sccp phones don't have two way speaker phone <br><br>b) is what I have, and are trying to get to work. see my previous email about the sip trunk. I don't know what to do to make unity go into the greeting for the user who was called.
<br><br>c) I'd like to, but I just switched voice mail for all my users, and I don't want to endure the nightmare of switching again. that's my long term goal.<br><br><div><span class="gmail_quote">On 1/24/06, <b class="gmail_sendername">
Michael J. Tubby B.Sc (Hons) G8TIC</b> <<a href="mailto:mike.tubby@thorcom.co.uk">mike.tubby@thorcom.co.uk</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div><font face="Arial" size="2">Options would appear to be:</font></div>
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2">a) use cheaper SCCP phones like 7905/7912 and stay
with CCM</font></div>
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2">b) put an asterisk box up and configure a SIP trunk
between CCM and Asterisk - I have done this and it works although there used to
be a bug with the CCM box not tearing down the RTP at the end of the call - it
appeared to rely on receiving an ICMP "port not reachable" from the other end -
this could probably be fixed with the appropriate rtptimeouts ?</font></div>
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2">You would add new users on Asterisk using SIP
phones and have a mixed system.</font></div>
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2">c) ditch the CCM and go 100% Asterisk</font></div>
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2">You might consider (b) in the short/medium with a
road-map towards (c)</font></div>
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2">Mike</font></div>
<div><font face="Arial" size="2"></font> </div>
<div><font face="Arial" size="2"></font> </div>
<blockquote style="border-left: 2px solid rgb(0, 0, 0); padding-right: 0px; padding-left: 5px; margin-left: 5px; margin-right: 0px;"><span class="q">
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;">----- Original Message ----- </div>
<div style="background: rgb(228, 228, 228) none repeat scroll 0% 50%; -moz-background-clip: -moz-initial; -moz-background-origin: -moz-initial; -moz-background-inline-policy: -moz-initial; font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;">
<b>From:</b>
<a title="sysread@gmail.com" href="mailto:sysread@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sys read</a> </div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><b>To:</b> <a title="asterisk-users@lists.digium.com" href="mailto:asterisk-users@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
Asterisk Users Mailing List -
Non-Commercial Discussion</a> </div></span><div><span class="e" id="q_108fd9eeaed881b9_2">
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><b>Sent:</b> Tuesday, January 24, 2006 3:56
PM</div>
<div style="font-family: arial; font-style: normal; font-variant: normal; font-weight: normal; font-size: 10pt; line-height: normal; font-size-adjust: none; font-stretch: normal;"><b>Subject:</b> Re: [Asterisk-Users] Asterisk
SIP phones to Cisco Unity via CCM4.0SIP Trunk</div>
<div><br></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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">kevin.ling@amphenol.com.tw</a>>
wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">Hi,</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" 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 color="#0000ff" face="Arial" size="2"></font></span> </div>
<div dir="ltr" align="left"><span></span><span></span><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div dir="ltr" align="left">
<hr>
<font face="Tahoma" size="2"><span><b>From:</b> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com
</a>[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">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>
<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 href="mailto:gary.richardson@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">gary.richardson@gmail.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">You
can run a SIP image on a 7940. Asterisk@home has pretty good<br>support
for it. Check the <a href="http://voip-info.org" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">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 href="mailto:sysread@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">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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</span></div><p>
</p><hr><span class="q">
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