<HTML><BODY style="word-wrap: break-word; -khtml-nbsp-mode: space; -khtml-line-break: after-white-space; ">CTI still does not support 2833, so if you have any CTI endpoints you're going to need MTP or DSP resource if low bitrate.<DIV><BR class="khtml-block-placeholder"></DIV><DIV>/Wes</DIV><DIV><BR><DIV><DIV>On Oct 13, 2006, at 8:20 PM, Zilk, David wrote:</DIV><BR class="Apple-interchange-newline"> <DIV class="Section1"><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> </DIV> <DIV><DIV><BR class="khtml-block-placeholder"></DIV><HR> This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. 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