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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Many of you have run across the problem of not being able to
utilize Caller ID Name (CNAM) information from a telco using H.323. Since
CNAM is provided by most telcos within the facility IE field, and this
isn’t supported in H.323, the solution has always been to convert to MGCP.
If you needed to stick with H.323 or gateway based call routing for
other reasons, such as needing to support other H.323 devices, or use TCL
scripts, you were out of luck.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>In one of those “why didn’t I think of that
before” moments, one of my colleagues thought to test connectivity using
an incoming SIP dial peer and SIP trunk on call manager. Lo and behold,
CNAM works perfectly! H.323 dial peers can still be used as needed to match
other inbound and outbound traffic, and since the routing decisions are being
done at the gateway, TCL scripts can still be used.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>A few SIP specific parameters are required on the gateway
including:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>signaling forward unconditional<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>sip<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'> bind all source-interface <interface><o:p></o:p></span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>!<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>!<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>sip-ua <o:p></o:p></span></font></p>
<p class=MsoNormal style='text-indent:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'> timers buffer-invite 5000<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>As well as adding “ session protocol sipv2 “ to
the incoming voip dial-peer.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Configuration on Call Manager includes adding an RFC 2833
DTMF-compliant MTP resource and setting up a SIP Trunk to the gateway.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>While we have not fully tested this configuration, it
appears to solve the problem of not being able to receive caller-id name
information while still maintaining gateway based call routing. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Can anyone point to any problems we might encounter with
this setup?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>David Zilk<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
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