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<DIV dir=ltr align=left><SPAN class=640200419-14102006><FONT face=Tahoma
size=2>Sounded like he was referring to a telco provided ISDN PRI terminating on
a Cisco VoIP gateway router, with SIP running between the customer owned gateway
and the CallManager.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=640200419-14102006><FONT face=Tahoma
size=2></FONT></SPAN><SPAN class=640200419-14102006><FONT face=Tahoma
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=640200419-14102006><FONT face=Tahoma
size=2>TELCO <--PRI--> VOIP GW <--SIP--> CM</FONT></SPAN></DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <B>On Behalf Of </B>Jason Aarons
(US)<BR><B>Sent:</B> Saturday, October 14, 2006 7:19 AM<BR><B>To:</B> Zilk,
David; cisco-voip@puck.nether.net<BR><B>Subject:</B> Re: [cisco-voip] TIP: CNAM
without requiring MGCP<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=Section1>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Whom was the telco
providing the SIP trunk?<o:p></o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial color=navy size=2><SPAN
style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<DIV>
<DIV class=MsoNormal style="TEXT-ALIGN: center" align=center><FONT
face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt">
<HR tabIndex=-1 align=center width="100%" SIZE=2>
</SPAN></FONT></DIV>
<P class=MsoNormal><B><FONT face=Tahoma size=2><SPAN
style="FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">From:</SPAN></FONT></B><FONT
face=Tahoma size=2><SPAN style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">
cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net]
<B><SPAN style="FONT-WEIGHT: bold">On Behalf Of </SPAN></B>Zilk,
David<BR><B><SPAN style="FONT-WEIGHT: bold">Sent:</SPAN></B> Friday, October 13,
2006 8:20 PM<BR><B><SPAN style="FONT-WEIGHT: bold">To:</SPAN></B>
cisco-voip@puck.nether.net<BR><B><SPAN
style="FONT-WEIGHT: bold">Subject:</SPAN></B> [cisco-voip] TIP: CNAM without
requiring MGCP</SPAN></FONT><o:p></o:p></P></DIV>
<P class=MsoNormal><FONT face="Times New Roman" size=3><SPAN
style="FONT-SIZE: 12pt"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; 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 face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; 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 face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; 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 face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal style="TEXT-INDENT: 0.5in"><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">signaling forward
unconditional<o:p></o:p></SPAN></FONT></P>
<P class=MsoNormal style="TEXT-INDENT: 0.5in"><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">sip<o:p></o:p></SPAN></FONT></P>
<P class=MsoNormal style="TEXT-INDENT: 0.5in"><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"> bind all source-interface
<interface><o:p></o:p></SPAN></FONT></P>
<P class=MsoNormal style="TEXT-INDENT: 0.5in"><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">!<o:p></o:p></SPAN></FONT></P>
<P class=MsoNormal style="TEXT-INDENT: 0.5in"><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">!<o:p></o:p></SPAN></FONT></P>
<P class=MsoNormal style="TEXT-INDENT: 0.5in"><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">sip-ua <o:p></o:p></SPAN></FONT></P>
<P class=MsoNormal style="TEXT-INDENT: 0.5in"><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"> timers buffer-invite
5000<o:p></o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; 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 face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; 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 face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; 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 face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; 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 face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">David
Zilk<o:p></o:p></SPAN></FONT></P>
<P class=MsoNormal><FONT face=Arial size=2><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"><o:p> </o:p></SPAN></FONT></P>
<DIV>
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