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<div class=Section1>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Is that only if they are running SIP loads or do SCCP phones
also support RFC2833?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Ryan Ratliff
[mailto:rratliff@cisco.com] <br>
<b>Sent:</b> Thursday, November 20, 2008 11:55 AM<br>
<b>To:</b> 'c3voip'; '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<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Lucida Sans","sans-serif";
color:blue'>Those phones have supported RFC2833 for a while now.</span><o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Lucida Sans","sans-serif";
color:blue'>If you want to check if your device supports RFC2833 reset the
phone and look at the CCM trace of it re-registering. </span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Lucida Sans","sans-serif";
color:blue'>You'll find this message:</span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Lucida Sans","sans-serif";
color:blue'>StationInit: (0000050) CapabilitiesRes capCount=12 caps= 25(40)
6(60) 4(40) 2(40) 7(60) 8(60) 86(60) 15(60) 16(60) 11(60) 12(60) 257(1).</span><o:p></o:p></p>
<p class=MsoNormal> <o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Lucida Sans","sans-serif";
color:blue'>The last item (257) indicates 2833 support.</span><o:p></o:p></p>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<p><span style='font-size:10.0pt'>-Ryan </span><o:p></o:p></p>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div class=MsoNormal align=center style='text-align:center'>
<hr size=2 width="100%" align=center>
</div>
<p class=MsoNormal style='margin-bottom:12.0pt'><b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> c3voip [mailto:c3voip@nc.rr.com] <br>
<b>Sent:</b> Thursday, November 20, 2008 11:09 AM<br>
<b>To:</b> 'Ryan Ratliff'; '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</span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>But I am guessing that our old 7940’s and 60’s
don’t…right?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>-C<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Ryan Ratliff
[mailto:rratliff@cisco.com] <br>
<b>Sent:</b> Thursday, November 20, 2008 10:52 AM<br>
<b>To:</b> 'c3voip'; '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<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Lucida Sans","sans-serif";
color:blue'>It can't, though most phones support RFC 2833 natively.</span><o:p></o:p></p>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<p><span style='font-size:10.0pt'>-Ryan </span><o:p></o:p></p>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div class=MsoNormal align=center style='text-align:center'>
<hr size=2 width="100%" align=center>
</div>
<p class=MsoNormal style='margin-bottom:12.0pt'><b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> 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</span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>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:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>-C<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;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:12.0pt'>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:12.0pt'><br>
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