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<BODY lang=EN-US vLink=purple link=blue><![if !supportLists]><![endif]><![if !supportLists]><![endif]><![if !supportLists]><![endif]><![if !supportLists]><![endif]>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff size=2>Hit 
another problem in my tutorial in converting over from Cisco CallManager to 
Asterisk. </FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff size=2>I have 
been following the&nbsp;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>&nbsp;on 
intergrating Asterisk and Cisco CallManager.&nbsp; </FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff size=2>I can 
make calls from CCM to Asterisk phones - and yes that felt good to get that 
working.</FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff size=2>My 
problem is that it does not work&nbsp;from the other direction.&nbsp; &nbsp;I 
cannot make calls from CCM phones to Asterisk Phones.&nbsp; </FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff size=2>I want 
to be able to dial 8 and the extension of the ccm phone.</FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff size=2>I am 
using CCM 3.3.(5) so I do not have the option to use a SIP turnk&nbsp;because it 
is not supported.&nbsp; I am also using h323 instead of ooh323.&nbsp; Not sure 
if that might make a difference.</FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff size=2>In 
Asterisk console I get:&nbsp;&nbsp;&nbsp; </FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff size=2>-- 
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>&nbsp;&nbsp;&nbsp; -- Requested transfer capability: 0x00 - 
SPEECH<BR>&nbsp;&nbsp;&nbsp; -- Called <A 
href="mailto:callman02/207@172.16.200.10:1720">callman02/207@172.16.200.10:1720</A><BR>&nbsp; 
== Everyone is busy/congested at this time (1:0/0/1)</FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff size=2>This 
is the contents of my h323.conf file:</FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2>=================</DIV>
<DIV></FONT></SPAN><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2>[general]<BR>port = 1720<BR>bindaddr = 172.17.100.2 </FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2>disallow=all<BR>allow=gsm&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
; Always allow GSM, it's cool 
:)<BR>allow=ulaw&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
; see doc/rtp-packetization for framing options<BR>allow=alaw&nbsp; 
</FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2>dtmfmode=rfc2833<BR>gatekeeper = 
DISABLE<BR>context=default<BR></FONT></SPAN><SPAN class=328482721-09062009><FONT 
face=Arial color=#0000ff 
size=2><BR>[callman02]<BR>type=friend<BR>context=default<BR>ip=172.16.200.10<BR>port=1720<BR>disallow=all<BR>allow=gsm&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<BR>allow=ulaw&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<BR>allow=alaw&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<BR>dtmfmode=rfc2833<BR>nat=no<BR>canreinvite=yes<BR>qualify=yes<BR></FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2>extensions.conf file</FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2>==============</FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2>[globals]<BR>CISCOTRUNK=H323/callman02<BR></FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2>[cisco]</FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff size=2>exten 
=&gt; _8XXX,1,Dial(${CISCOTRUNK}/${EXTEN:1}@172.16.200.10:1720)<BR>exten =&gt; 
_8XXX,n,Congestion()<BR>exten =&gt; _8XXX,n,Hangup()<BR></FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff size=2>Jimmy 
Ezell</FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2>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></FONT></SPAN></DIV>
<DIV><SPAN class=328482721-09062009><FONT face=Arial color=#0000ff 
size=2>&nbsp;</DIV></FONT></SPAN></BODY></HTML>