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<DIV dir=ltr align=left><SPAN class=294444917-07032011><FONT face=Arial
color=#0000ff size=2>Hi Mike,</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=294444917-07032011><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=294444917-07032011><FONT face=Arial color=#0000ff size=2>Some
remote PBXs use a SETUP message with an enclosed Facility Information element.
Buffering refers to those who instead send a completely separate FACILITY
message containing the name information to display. The gateway would have to
buffer (in Cisco parlance) for a few milliseconds to collate the SETUP
and the calling name information. Your choice depends on what your provider's
equipment sends to you.</FONT></SPAN></DIV>
<DIV><SPAN class=294444917-07032011><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=294444917-07032011></SPAN><SPAN
class=294444917-07032011></SPAN><SPAN class=294444917-07032011></SPAN><SPAN
class=294444917-07032011><FONT face=Arial color=#0000ff size=2>I don't have
personal experience of using this command, but I would lean towards applying it
globally at the "voice service voip" level if you know you want to use it on
every T1/E1 circuit. Otherwise, I presume the voice class would need to be
associated with your incoming pots dial-peer, but this doesn't seem to be
documented.</FONT></SPAN></DIV>
<DIV><SPAN class=294444917-07032011><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=294444917-07032011><FONT face=Arial color=#0000ff size=2>Hope
this helps</FONT></SPAN></DIV>
<DIV><SPAN class=294444917-07032011><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=294444917-07032011><FONT face=Arial color=#0000ff
size=2>James.</FONT></SPAN></DIV>
<DIV><SPAN class=294444917-07032011><FONT face=Arial color=#0000ff
size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left>
<HR tabIndex=-1>
</DIV>
<DIV dir=ltr align=left><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>Mike King<BR><B>Sent:</B> 07 March 2011 15:57<BR><B>To:</B>
Cisco VoIPoE List<BR><B>Subject:</B> [cisco-voip] Configuring H.323 Calling Name
Display<BR></FONT><BR></DIV>
<DIV></DIV>I'm looking to configure H323 Calling Name Display..
<DIV><BR></DIV>
<DIV><A
href="http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-cm.html#wp1324664">http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb-gw-cm.html#wp1324664</A></DIV>
<DIV><BR></DIV>
<DIV>It provides 4 ways to configure this:</DIV>
<DIV><BR></DIV>
<DIV>Voice Class Level - With Buffering</DIV>
<DIV>Voice Class Level - Without Buffering</DIV>
<DIV>Voice Service Level - With Buffering</DIV>
<DIV>Voice Service Level - Without Buffering</DIV>
<DIV><BR></DIV>
<DIV>I have a 2921, with a single PRI providing service to this branch office
(Also providing SRST)</DIV>
<DIV><BR></DIV>
<DIV>My question, what's the difference between buffering and without
buffering?</DIV>
<DIV>also,</DIV>
<DIV>Would I have to configure it at both the Voice Class and Service level, or
just one of the two? If it's just the one, I'm assuming the Voice Service
level would make the most sense for a single T1 type environment.</DIV>
<DIV><BR></DIV>
<DIV>Mike </DIV>
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