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<div class=Section1>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>H323 does support calling party number,
just not name. Like Jason mentioned before, the options are there and you can
see name traverse inbound in debugs, but the gateway does not send it on to
CallManager.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<div>
<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span style='font-size:12.0pt;font-family:"Times New Roman"'>
<hr size=2 width="100%" align=center tabindex=-1>
</span></font></div>
<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-weight:bold'>From:</span></font></b><font size=2><span style='font-size:
10.0pt'> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <b><span style='font-weight:bold'>On
Behalf Of </span></b>Jason Burwell<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, August 30, 2006
1:36 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> Re: [cisco-voip] MGCP vs
H323</span></font><font face="Times New Roman"><span style='font-family:"Times New Roman"'><o:p></o:p></span></font></p>
</div>
<p class=MsoNormal><font size=3 face=Tahoma><span style='font-size:12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>We have just purchased CVP and are being
told that we can no longer use MGCP gateways and must migrate to H323.
According to what I have heard here, H323 will not support Caller Name but will
H323 support Calling Party Number capability?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Jason<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<div>
<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span style='font-size:12.0pt;font-family:"Times New Roman"'>
<hr size=2 width="100%" align=center tabindex=-1>
</span></font></div>
<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-weight:bold'>From:</span></font></b><font size=2><span style='font-size:
10.0pt'> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <b><span style='font-weight:bold'>On
Behalf Of </span></b><st1:PersonName w:st="on">Jason Aarons</st1:PersonName>
(US)<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, August 30, 2006
12:52 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> Re: [cisco-voip] MGCP vs
H323</span></font><font face="Times New Roman"><span style='font-family:"Times New Roman"'><o:p></o:p></span></font></p>
</div>
<p class=MsoNormal><font size=3 face=Tahoma><span style='font-size:12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I believe the issue is with CallManager
not supporting both types of facility ie as you can see the Calling Party Name
in the router’s debug isdn q931.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<div>
<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span style='font-size:12.0pt;font-family:"Times New Roman"'>
<hr size=2 width="100%" align=center tabindex=-1>
</span></font></div>
<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-weight:bold'>From:</span></font></b><font size=2><span style='font-size:
10.0pt'> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <b><span style='font-weight:bold'>On
Behalf Of </span></b>Linsemier, Matthew<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, August 30, 2006
12:11 PM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Matt
Slaga</st1:PersonName> (US); Joe Pollere (US); Nick Kassel;
cisco-voip@puck.nether.net<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [cisco-voip] MGCP vs
H323</span></font><font face="Times New Roman"><span style='font-family:"Times New Roman"'><o:p></o:p></span></font></p>
</div>
<p class=MsoNormal><font size=3 face=Tahoma><span style='font-size:12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Calling Name/Number is one of the main
reasons we utilize MGCP over H.323. This coupled with failover (which
seems to be handled much more gracefully then in an H.323 environment) keep us
on this protocol. If H.323 could handle Calling Name, I would definitely
make the move. It would certainly make fax integration with products such
as Unity or XMedius fax much easier. I, however, find myself in a
position where I couldn’t possibly take away features that my users have
come to expect and love. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I did hear rumblings that H.323 was going
to support Calling Name in a later IOS release. The information is there
on the router, just not getting processed. Can any of you Cisco folks
comment (off the record)?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>-Matt<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<div>
<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span style='font-size:12.0pt;font-family:"Times New Roman"'>
<hr size=2 width="100%" align=center tabindex=-1>
</span></font></div>
<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-weight:bold'>From:</span></font></b><font size=2><span style='font-size:
10.0pt'> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <b><span style='font-weight:bold'>On
Behalf Of </span></b><st1:PersonName w:st="on">Matt Slaga</st1:PersonName> (US)<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, August 30, 2006
12:00 PM<br>
<b><span style='font-weight:bold'>To:</span></b> Joe Pollere (US); Nick Kassel;
cisco-voip@puck.nether.net<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [cisco-voip] MGCP vs
H323</span></font><font face="Times New Roman"><span style='font-family:"Times New Roman"'><o:p></o:p></span></font></p>
</div>
<p class=MsoNormal><font size=3 face=Tahoma><span style='font-size:12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>At this point, MGCP is able to provide
name/number display on a PRI (if provided from the carrier) whereas this is not
yet possible in H323.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I have started to hear some rumblings
within Cisco about MGCP being slowly phased out and eventually being replaced
with SIP and/or H323. Sounds about right since Cisco has been shoving
MGCP down everyone’s throat for the past three years that they would want
to keep this down to a rumbling.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Personally, I would only use MGCP when
interconnecting with a PBX during a migration to keep from having thousands of
dial peers. Or, if I needed name/number display on a PRI.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Funny thing is (and this may be fixed
now), but with FXO-M1s the only way to get name/number display was with
H323. MGCP couldn’t do it. Sounds kind of backwards that
it’s just the opposite with a PRI<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<div>
<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span style='font-size:12.0pt;font-family:"Times New Roman"'>
<hr size=2 width="100%" align=center tabindex=-1>
</span></font></div>
<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-weight:bold'>From:</span></font></b><font size=2><span style='font-size:
10.0pt'> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <b><span style='font-weight:bold'>On
Behalf Of </span></b>Joe Pollere (US)<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, August 30, 2006
10:51 AM<br>
<b><span style='font-weight:bold'>To:</span></b> Nick Kassel;
cisco-voip@puck.nether.net<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [cisco-voip] MGCP vs
H323</span></font><font face="Times New Roman"><span style='font-family:"Times New Roman"'><o:p></o:p></span></font></p>
</div>
<p class=MsoNormal><font size=3 face=Tahoma><span style='font-size:12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Z@R44A.tmp"><span
style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Nick,<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Z@R44A.tmp"><span
style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Z@R44A.tmp"><span
style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>This is from the GWGK
course book:<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><b><font size=2 face="Z@R44A.tmp"><span
style='font-size:11.0pt;font-family:"Z\@R44A\.tmp";font-weight:bold'><o:p> </o:p></span></font></b></p>
<p class=MsoNormal style='text-autospace:none'><b><font size=2 face="Z@R44A.tmp"><span
style='font-size:11.0pt;font-family:"Z\@R44A\.tmp";font-weight:bold'>Using H.323
as the call control protocol to a gateway has the following advantages:<o:p></o:p></span></font></b></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Z@R44A.tmp"><span
style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'><o:p> </o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>H.323
provides integrated access. Data and voice channels can be placed on the
same T1.<o:p></o:p></span></font></li>
</ul>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>For
example, for a service provider like AT&T, FR and PRI can be placed on the
same T1.<o:p></o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>H.323
provides support for fractional PRI.<o:p></o:p></span></font></li>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Gateways
support caller ID on FXO ports. CallManager does not support caller ID on
FXO<o:p></o:p></span></font></li>
</ul>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>ports
from MGCP gateways.<o:p></o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmpOOEnc"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmpOOEnc"'>Many more TDM interface types and signaling protocols</span></font><font
size=2 face=Arial><span style='font-size:11.0pt;font-family:Arial'>—</span></font><font
size=2 face="Z@R44A.tmpOOEnc"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmpOOEnc"'>for example, analog-Direct<o:p></o:p></span></font></li>
</ul>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Inward
Dialed (DID), recEive and transMit (E&M), T1 Feature Group-D (FGD), and E1<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmpOOEnc"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmpOOEnc"'>R2</span></font><font
size=2 face=Arial><span style='font-size:11.0pt;font-family:Arial'>—</span></font><font
size=2 face="Z@R44A.tmpOOEnc"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmpOOEnc"'>can
be used.<o:p></o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>H.323
drops DSPs on hairpinned calls to enable capabilities like ISDN video
switching.<o:p></o:p></span></font></li>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Gateway
resident applications like Toolkit Command Language (TCL) and voice<o:p></o:p></span></font></li>
</ul>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>extensible
markup language (VXML) can be used. TCL and VXML applications provide<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>IVR
features and call control functionality such as call forwarding, conference
calling, and<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>voice
mail.<o:p></o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>CAC
network design with H.323 gatekeepers is often necessary when voice and
video<o:p></o:p></span></font></li>
</ul>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>coexist
in a network and Cisco CallManager is not the only call controller in the
network.<o:p></o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>There
are no release dependencies between gateways and Cisco CallManager for<o:p></o:p></span></font></li>
</ul>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>supporting
new voice hardware. New hardware cards on Cisco IOS gateways become<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>immediately
available for use with all existing Cisco CallManager releases.<o:p></o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>H.323
enables a much easier migration architecture to SIP because the
fundamental<o:p></o:p></span></font></li>
</ul>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmpOOEnc"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmpOOEnc"'>concepts
of H.323 and SIP</span></font><font size=2 face=Arial><span style='font-size:
11.0pt;font-family:Arial'>—</span></font><font size=2
face="Z@R44A.tmpOOEnc"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmpOOEnc"'>for
example, distributed control with dial-peer<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmpOOEnc"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmpOOEnc"'>configurations</span></font><font
size=2 face=Arial><span style='font-size:11.0pt;font-family:Arial'>—</span></font><font
size=2 face="Z@R44A.tmpOOEnc"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmpOOEnc"'>are
the same.<o:p></o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Calls
from IP phones through an H323 gateway are dropped on a CallManager
failover<o:p></o:p></span></font></li>
</ul>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>unless
SRST mode is enabled. With SRST enabled, the calls are preserved.</span></font><font
size=2 face="Z@R44A.tmp"><span style='font-size:10.0pt;font-family:"Z\@R44A\.tmp"'><o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><b><font size=2 face="Z@R44A.tmp"><span
style='font-size:11.0pt;font-family:"Z\@R44A\.tmp";font-weight:bold'>Using MGCP
as the call control protocol to a gateway has the following advantages:<o:p></o:p></span></font></b></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Z@R44A.tmp"><span
style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'><o:p> </o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Centralized
configuration, control, and download from Cisco CallManager<o:p></o:p></span></font></li>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Better
feature interaction with capabilities like caller ID and name display<o:p></o:p></span></font></li>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Easy,
centralized dial-plan management<o:p></o:p></span></font></li>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Gateway
voice security features (voice encryption) as of Cisco IOS Software
Release<o:p></o:p></span></font></li>
</ul>
<p class=MsoNormal style='margin-left:.25in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>12.3.(5</span></font><font
size=1 face="Z@R44A.tmp"><span style='font-size:7.0pt;font-family:"Z\@R44A\.tmp"'>th</span></font><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>)T<o:p></o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Q
Signaling (QSIG) supplementary services as supported by Cisco CallManager:<o:p></o:p></span></font></li>
<ul style='margin-top:0in' type=circle>
<li class=MsoNormal style='mso-list:l0 level2 lfo1;text-autospace:none'><font
size=2 face=Arial><span style='font-size:11.0pt;font-family:Arial'>—</span></font><font
size=2 face="Z@R44A.tmpOOEnc"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmpOOEnc"'> </span></font><font size=2 face="Z@R44A.tmp"><span
style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Cisco CallManager
interconnects to a QSIG network using an MGCP gateway and<o:p></o:p></span></font></li>
</ul>
</ul>
<p class=MsoNormal style='margin-left:.5in;text-indent:.5in;text-autospace:
none'><font size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmp"'>T1 or E1 PRI connections to a private integrated services
network (PISN). The<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-indent:.5in;text-autospace:
none'><font size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmp"'>MGCP gateway establishes the call connections. Using the PRI
backhaul<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-indent:.5in;text-autospace:
none'><font size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmp"'>mechanism, the gateway passes the QSIG messages to the Cisco
CallManager to set<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-indent:.5in;text-autospace:
none'><font size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmp"'>up QSIG calls and send QSIG messages to control features.<o:p></o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<ul style='margin-top:0in' type=circle>
<li class=MsoNormal style='mso-list:l0 level2 lfo1;text-autospace:none'><font
size=2 face=Arial><span style='font-size:11.0pt;font-family:Arial'>—</span></font><font
size=2 face="Z@R44A.tmpOOEnc"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmpOOEnc"'> </span></font><font size=2 face="Z@R44A.tmp"><span
style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>When a PBX is
connected to a gateway that is using QSIG via H.323, calls that are<o:p></o:p></span></font></li>
</ul>
</ul>
<p class=MsoNormal style='margin-left:.5in;text-indent:.5in;text-autospace:
none'><font size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmp"'>made between phones on the PBX and IP phones attached to the
Cisco CallManager<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-indent:.5in;text-autospace:
none'><font size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmp"'>can have only basic PRI functionality. The gateway that
terminates the QSIG<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-indent:.5in;text-autospace:
none'><font size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmp"'>protocol provides only the calling line ID (CLID) and DID
number, instead of Cisco<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-indent:.5in;text-autospace:
none'><font size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmp"'>CallManager providing that information.<o:p></o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<li class=MsoNormal style='mso-list:l0 level1 lfo1;text-autospace:none'><font
size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Enhanced
call survivability:<o:p></o:p></span></font></li>
<ul style='margin-top:0in' type=circle>
<li class=MsoNormal style='mso-list:l0 level2 lfo1;text-autospace:none'><font
size=2 face=Arial><span style='font-size:11.0pt;font-family:Arial'>—</span></font><font
size=2 face="Z@R44A.tmpOOEnc"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmpOOEnc"'> </span></font><font size=2 face="Z@R44A.tmp"><span
style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Calls from IP phones
through an MGCP gateway are preserved on a CallManager<o:p></o:p></span></font></li>
</ul>
</ul>
<p class=MsoNormal style='margin-left:.75in;text-indent:.25in;text-autospace:
none'><font size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmp"'>failover. This feature avoids dropped calls when applying the
monthly operating<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-indent:.5in;text-autospace:
none'><font size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmp"'>system service release on the Cisco CallManagers<o:p></o:p></span></font></p>
<ul style='margin-top:0in' type=disc>
<ul style='margin-top:0in' type=circle>
<li class=MsoNormal style='mso-list:l0 level2 lfo1;text-autospace:none'><font
size=2 face=Arial><span style='font-size:11.0pt;font-family:Arial'>—</span></font><font
size=2 face="Z@R44A.tmpOOEnc"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmpOOEnc"'> </span></font><font size=2 face="Z@R44A.tmp"><span
style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>In SRST mode, calls
from IP phones through an MGCP gateway are preserved on<o:p></o:p></span></font></li>
</ul>
</ul>
<p class=MsoNormal style='margin-left:.5in;text-indent:.5in;text-autospace:
none'><font size=2 face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:
"Z\@R44A\.tmp"'>MGCP fallback for calls on analog or CAS circuits. Calls on
ISDN circuits are<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:1.0in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>dropped
on fallback.<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:1.0in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>HTH’s<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-autospace:none'><font size=2
face="Z@R44A.tmp"><span style='font-size:11.0pt;font-family:"Z\@R44A\.tmp"'>Joe<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<div>
<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span style='font-size:12.0pt;font-family:"Times New Roman"'>
<hr size=2 width="100%" align=center tabindex=-1>
</span></font></div>
<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-weight:bold'>From:</span></font></b><font size=2><span style='font-size:
10.0pt'> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <b><span style='font-weight:bold'>On
Behalf Of </span></b>Nick Kassel<br>
<b><span style='font-weight:bold'>Sent:</span></b> Wednesday, August 30, 2006
10:35 AM<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] MGCP vs H323</span></font><font
face="Times New Roman"><span style='font-family:"Times New Roman"'><o:p></o:p></span></font></p>
</div>
<p class=MsoNormal><font size=3 face=Tahoma><span style='font-size:12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>Does anyone know if there is a document that lists
the differences or benefits of using either MGCP or H323.</span></font><span
lang=EN-GB><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'> </span></font><span lang=EN-GB><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>I need to install a gateway for PSTN breakout in a DR
site so it won’t be needed for redundancy or anything like that. </span></font><span
lang=EN-GB><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'> </span></font><span lang=EN-GB><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>All our branch offices currently use H323 and we have
H323 for our HQ so that they can be used for redundancy should be an issue with
the local branch PSTN. </span></font><span lang=EN-GB><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'> </span></font><span lang=EN-GB><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>Just wondering whether it might be better to go with
MGCP in this instance.</span></font><span lang=EN-GB><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'> </span></font><span lang=EN-GB><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>Is it easier to mask the calling numbers with MGCP?</span></font><span
lang=EN-GB><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'> </span></font><span lang=EN-GB><o:p></o:p></span></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'> </span></font><span lang=EN-GB><o:p></o:p></span></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span lang=EN-GB style='font-size:12.0pt;font-family:
"Times New Roman"'><o:p> </o:p></span></font></p>
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