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<DIV dir=ltr align=left><SPAN class=148255115-20112008><FONT face="Lucida Sans"
color=#0000ff size=2>It can't, though most phones support RFC 2833
natively.</FONT></SPAN></DIV>
<DIV><FONT face="Lucida Sans" color=#0000ff size=2></FONT> </DIV><!-- Converted from text/plain format -->
<P><FONT size=2>-Ryan </FONT></P>
<DIV> </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>c3voip<BR><B>Sent:</B> Thursday, November 20, 2008 10:49 AM<BR><B>To:</B>
'Jason Burns'<BR><B>Cc:</B> cisco-voip@puck.nether.net<BR><B>Subject:</B> Re:
[cisco-voip] SIP audio conference bridge to CCM 4.1.3<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=Section1>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'">How
can CUCM 6.X or 7.X translate SCCP DTMF from out-of-path to in-path for SIP
without MTP’s? How does that work?<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'"><o:p> </o:p></SPAN></P>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'">-C<o:p></o:p></SPAN></P>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'"><o:p> </o:p></SPAN></P>
<P class=MsoNormal><SPAN
style="FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: 'Calibri','sans-serif'"><o:p> </o:p></SPAN></P>
<DIV
style="BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: medium none; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P class=MsoNormal><B><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">From:</SPAN></B><SPAN
style="FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'"> Jason Burns
[mailto:burns.jason@gmail.com] <BR><B>Sent:</B> Wednesday, November 19, 2008
7:21 PM<BR><B>To:</B> c3voip<BR><B>Cc:</B>
cisco-voip@puck.nether.net<BR><B>Subject:</B> Re: [cisco-voip] SIP audio
conference bridge to CCM 4.1.3<o:p></o:p></SPAN></P></DIV>
<P class=MsoNormal><o:p> </o:p></P>
<P class=MsoNormal style="MARGIN-BOTTOM: 12pt">A few options:<BR><BR>1. Upgrade
to CCM 6.Xor 7.X so you don't need the MTP anymore.<BR>2. Provision a separate
CallManager (or 2) for the sole purpose of acting as an MTP and increase the
number of possible channels on it.<BR>3. Purchase some CMMs<BR>4. Purchase some
3845s loaded with DSPs.<BR><BR>Options 2, 3, and 4 don't really make much sense.
As soon as you had CMMs, 3845s, or extra IPVMSApp CallManagers you wouldn't
really need your SIP conference solution anymore.<o:p></o:p></P>
<DIV>
<P class=MsoNormal>On Wed, Nov 19, 2008 at 3:46 PM, c3voip <<A
href="mailto:c3voip@nc.rr.com">c3voip@nc.rr.com</A>> wrote:<o:p></o:p></P>
<DIV>
<DIV>
<P>I am considering a purchase of a SIP audio conference bridge to connect via a
SIP trunk from CCM <A href="http://4.1.3." target=_blank>4.1.3.</A>
<o:p></o:p></P>
<P> <o:p></o:p></P>
<P>Since the SIP trunk requires an MTP, what could I use for MTP resources to
get up to 500 SCCP G.711 calls across my SIP trunk? From what I have read
the CCM software MTP's are only good for 48 calls, what would be a better
solution?<o:p></o:p></P>
<P> <o:p></o:p></P>
<P>Thanks,<o:p></o:p></P>
<P>-C<o:p></o:p></P>
<P> <o:p></o:p></P></DIV></DIV>
<P class=MsoNormal
style="MARGIN-BOTTOM: 12pt"><BR>_______________________________________________<BR>cisco-voip
mailing list<BR><A
href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</A><BR><A
href="https://puck.nether.net/mailman/listinfo/cisco-voip"
target=_blank>https://puck.nether.net/mailman/listinfo/cisco-voip</A><o:p></o:p></P></DIV>
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