what you are reading on Cisco manual "DN" is a completely different concept that what we are dealing in asterisk.<br>In CME you refer to each number as a DN, that concept does not exist on Asterisk.<br>Although Asterisk support SCCP (Skinny) and H323, but its always easier and better to use SIP or IAX.
<br>if you like to have a reception phone with BLF, there are lots of options to choose from.<br>Beside the fact that i don't like quality of Cisco Phones, I usually get better and professional results with Polycom.<br>
But again that is my opinion.<br><br><br><div><span class="gmail_quote">On 8/6/07, <b class="gmail_sendername">Ryan Amos</b> <<a href="mailto:ramos@finetooth.com">ramos@finetooth.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;">
<div vlink="purple" link="blue" lang="EN-US">
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">The 7914 only works under SCCP; the SIP firmware does not
support it at all (the expansion panel won't even power on fully.) The SCCP
channel driver under Asterisk doesn't really support the 7914 very well,
currently it will only show onhook/offhook state (though there has been much
discussion recently about changing this.) If you want to do this with
SIP then you're better off with something like the grandstream mentioned, or
just use the Flash Operator Panel (IMO it gives you more flexibility at a much
lower cost.)</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">I have personally found "receptionist phone" functionality
handled much better with FOP. I have a 7914 and its functionality (and
usefulness) is very limited under Asterisk.</font></span></div><br>
<div dir="ltr" align="left" lang="en-us">
<hr>
<font face="Tahoma" size="2"><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>James R.
Stevens<br><b>Sent:</b> Monday, August 06, 2007 10:41 AM</font><div><span class="e" id="q_1143c0678db5dd48_1"><font face="Tahoma" size="2"><br><b>To:</b> Asterisk
Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re:
[asterisk-users] Learn some terminalogy before
mountingthistask.<br></font></span></div><br></div><div><span class="e" id="q_1143c0678db5dd48_3">
<div></div>
<div>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Thank
you for your reply as it is exactly what we would need. Sorry I didn't find it
myself. I do have a question about configuration within Asterisk.
</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">I'm
reading the PDF on the Cisco Expansion module and it says 'When used as a DN key
buttons are illuminated …'</span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p><span style="font-size: 11pt; color: rgb(31, 73, 125);">Is
that what we are doing within Asterisk or Trixbox when we configure an
extension? (A Directory Number??)</span></p>
<p><a name="1143c0678db5dd48__MailEndCompose"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></a></p>
<div style="border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; border-width: 1pt medium medium; padding: 3pt 0in 0in;">
<p><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;">
<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>James
FitzGibbon<br><b>Sent:</b> Monday, August 06, 2007 7:37 AM<br><b>To:</b>
Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re:
[asterisk-users] Learn some terminalogy before
mountingthistask.</span></p></div>
<p> </p>
<p>On 8/5/07, <b>James R. Stevens</b> <<a href="mailto:jstevens@athensdistributing.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">jstevens@athensdistributing.com</a>>
wrote:</p>
<div>
<blockquote style="border-style: none none none solid; border-color: -moz-use-text-color -moz-use-text-color -moz-use-text-color rgb(204, 204, 204); border-width: medium medium medium 1pt; padding: 0in 0in 0in 6pt; margin-left: 4.8pt; margin-right: 0in;">
<p>In the design of an Asterisk system using Cisco 7900 series
SIP phones<br>we are struggling with giving the reception folks (3) hardware
that can<br>tell them the status of everyone in the office (10 or so) (On the
phone, <br>out of office etc) Something that would register each of the
extensions<br>we choose and give status of that ext.<br><br>What hardware
(Phone or other) could we give the receptionist to
do<br>this?</p></blockquote>
<div>
<p><br>You're probably looking for something like
this:<br><br><a href="http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008883d.html" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008883d.html
</a><br><br>I have no experience integrating this specific piece of hardware
with Asterisk, but I've done what you're trying to do with the Grandstream
equivalent for our front reception:<br><br><a href="http://www.grandstream.com/gxp2000.html" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://www.grandstream.com/gxp2000.html</a><br><br>and<br>
<br><a href="http://www.grandstream.com/gxp2000ext.html" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://www.grandstream.com/gxp2000ext.html</a><br><br>As
I understand it, so long as the device can do a SIP SUBSCRIBE for each extension
you want to monitor and you configure hints in your Asterisk dialplan for those
extensions, it should work. You may need to set 'subscribecontext' (in
sip.conf) for the phone that will be watching the extensions unless your hints
are in the same context as the phone uses for outbound dialing.<br><br>Of
course, what the device does with the various payloads contained in the SIP
NOTIFY messages is going to be different for each phone. On the
Grandstream I can see 'not in use' (solid green) and 'ringing' (flashing red)
distinctly, but 'unavailable' and 'in use' are both mapped to a solid red, which
makes it somewhat useless for transiently connected user agents like softphones.
</p></div></div>
<p><br>Hopefully someone with experience will speak up and
confirm that the 7900 series does interop properly with Asterisk for SUBSCRIBE
and NOTIFY.<br><br>If that doesn't work, you could always go with a software
solution, like the Flash Operator Panel. voip-info has a list (look at the
"Operator" section on the page): <a href="http://www.voip-info.org/wiki/view/Asterisk+GUI" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://www.voip-info.org/wiki/view/Asterisk+GUI</a>
<br><br>--
<br>j. </p></div><br>-- <br>This message has been scanned for viruses
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