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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The features you mentioned will work fine, but you’ll need to
also have to maintain a tftp server to provision the phones and ensure a quick
boot. In my experience, the only annoying thing about sip loads on Cisco phones
is that they don’t support sidecars for admin.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><a href="http://www.voip-info.org/wiki-Setup+SiP+on+7940+-+7960">http://www.voip-info.org/wiki-Setup+SiP+on+7940+-+7960</a><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Make sure you remove any callmanager related info from your DHCP
scope before you deploy Asterisk if they previously had a callmanager
installed. When completing these types of conversions, you run the risk of the
phones going to an unprovisioned state if they start trying to access a
callmanager that has been removed from the network. It sucks to get called back
to a job a few weeks later when the customer’s phone gets whacked after it was
unplugged and rebooted.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>Good luck<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:#1F497D'>--------------------------------------------------<br>
Salvatore Giudice<br>
Salvatore.Giudice@VoIPSecurityTraining.com<br>
<br>
VoIP Security Training, LLC<br>
http://VoIPSecurityTraining.com<br>
<br>
848 N. Rainbow Blvd. #1676<br>
Las Vegas, NV 89107<br>
Phone: (617) 959-7625<br>
Fax: (214) 279-2906<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Erick
Perez<br>
<b>Sent:</b> Thursday, May 03, 2007 8:33 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> [asterisk-users] Asterisk 1.4 and Cisco Phones 7940<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>I have read the wiki and several other internet documents.
Can anyone make a comment as to what kind of functionality will you loose if
you use Cisco 7940 phones with asterisk 1.4<br>
things like: MWI, call transfer, conference,etc,etc. <br>
I have a customer with 6 of those phones that he like to use with the asteirsk
PBX.<br>
<br>
thanks,<br>
<br clear=all>
<br>
-- <br>
------------------------------------------------------------<br>
Erick Perez<br>
------------------------------------------------------------ <o:p></o:p></p>
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