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<P><SPAN class=932163218-28122004><FONT size=2>I have a similar
setup. To make it easy and get the best of both worlds, have the
Linux softphones (SIP or IAX) register to Asterisk. Keep the
physical phones registered to CM. From there setup a dialplan on
both Call Manager and Asterisk to relay calls between the two
systems. For example, assign all physical phones extension 2XXX and
softphones 3XXX. Have asterisk route 2XXX calls to CM via SIP and
vice versa on Call Manager.</FONT></SPAN></P>
<P><SPAN class=932163218-28122004><FONT size=2>Also, just so that you are aware
you can register a SIP Linux softclient to Cisco Call Manager if you are running
Version 4.1</FONT></SPAN></P>
<P><SPAN class=932163218-28122004><FONT
size=2>-----------------------------------------------</FONT></SPAN></P>
<P><FONT size=2>Hello everybody,</FONT></P>
<P><FONT size=2>im newbie in VoIP, but find this project asterisk very
interesting, i tried to install and its a great sw, i really get sorprised about
all of its functions, we need to use the asterisk server in conjunction with
cisco callmanager.</FONT></P>
<P><FONT size=2>We have a Cisco Callmanager 4.1 and the clients are softphones
from cisco IPCommunicator, but all the support service of our company are linux
machines, i read about callmanager uses skinny a propetary protocol and there
are no softphones from linux to talk with it, so we need to install vmware to
use ipcommunicator or the other solutions as i read is get the asterisk server
using sip phones in the linux and windows machines and configure the call
manager to talk with the asterisk server thru sip protocol, is this the real way
to do that?? is there a easy way to do this?? i found this link</FONT></P>
<P><A
href="http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration"><U><FONT
color=#0000ff><FONT
size=2>http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration</FONT></U></FONT></A></P>
<P><FONT size=2>but i need to know what things to do to transfer all the
extensions from de callmanager to the asterisk sw, or if only made the changes
in the sip.conf as said in the link above the callmanager gets all the
control??</FONT></P>
<P><FONT size=2>or if i need to declare all the extensions in the asterisk?? can
anybody help me??</FONT></P>
<P><FONT size=2>TIA</FONT></P>
<P><FONT size=2>Edgar</FONT></P></DIV>
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