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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Make sure you are stripping the 8 on inbound calls to that H323
gateway<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>under CCM and that it has a valid search space to find your
extensions...<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>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal style='margin-left:.5in'><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>Jimmy
Ezell<br>
<b>Sent:</b> Tuesday, June 09, 2009 3:13 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> [asterisk-users] Asterisk to CCM<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal style='margin-left:.5in'><o:p> </o:p></p>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>Hit another problem in my tutorial
in converting over from Cisco CallManager to Asterisk. </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>I have been following
the instructions at : <a
href="http://voip-info.capatres.com/wiki/view/Asterisk+Cisco+CallManager+Integration.html">http://voip-info.capatres.com/wiki/view/Asterisk+Cisco+CallManager+Integration.html</a> on
intergrating Asterisk and Cisco CallManager. </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>I can make calls from CCM to
Asterisk phones - and yes that felt good to get that working.</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>My problem is that it does not
work from the other direction. I cannot make calls from CCM
phones to Asterisk Phones. </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>I want to be able to dial 8 and
the extension of the ccm phone.</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>I am using CCM 3.3.(5) so I do not
have the option to use a SIP turnk because it is not supported. I am
also using h323 instead of ooh323. Not sure if that might make a
difference.</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>In Asterisk console I
get: </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>-- Executing [8207@internal:1]
Dial("SIP/207-08bd64c8", "<a
href="mailto:H323/callman02/207@172.16.200.10:1720">H323/callman02/207@172.16.200.10:1720</a>")
in new stack<br>
-- Requested transfer capability: 0x00 - SPEECH<br>
-- Called <a href="mailto:callman02/207@172.16.200.10:1720">callman02/207@172.16.200.10:1720</a><br>
== Everyone is busy/congested at this time (1:0/0/1)</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>This is the contents of my
h323.conf file:</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>=================<o:p></o:p></span></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>[general]<br>
port = 1720<br>
bindaddr = 172.17.100.2 </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>disallow=all<br>
allow=gsm
; Always allow GSM, it's cool :)<br>
allow=ulaw
; see doc/rtp-packetization for framing options<br>
allow=alaw </span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>dtmfmode=rfc2833<br>
gatekeeper = DISABLE<br>
context=default<br>
<br>
[callman02]<br>
type=friend<br>
context=default<br>
ip=172.16.200.10<br>
port=1720<br>
disallow=all<br>
allow=gsm <br>
allow=ulaw <br>
allow=alaw <br>
dtmfmode=rfc2833<br>
nat=no<br>
canreinvite=yes<br>
qualify=yes</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>extensions.conf file</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>==============</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>[globals]<br>
CISCOTRUNK=H323/callman02</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>[cisco]</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>exten =>
_8XXX,1,Dial(${CISCOTRUNK}/${EXTEN:1}@172.16.200.10:1720)<br>
exten => _8XXX,n,Congestion()<br>
exten => _8XXX,n,Hangup()</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>Jimmy Ezell</span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'>Converting CCM to Asterisk <a
href="http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html">http://qvlweb.blogspot.com/2009/04/asterisk-pbx-install-index.html</a></span><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-left:.5in'><span style='font-size:10.0pt;
font-family:"Arial","sans-serif";color:blue'> <o:p></o:p></span></p>
</div>
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